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gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.

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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
This commit is contained in:
Wim Taymans 2008-09-05 13:52:34 +00:00
parent 64cd01e7e8
commit a35d1dde42
14 changed files with 277 additions and 107 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

59
ChangeLog
View file

@ -1,3 +1,62 @@
2008-09-05 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
2008-09-05 Zaheer Abbas Merali <zaheerabbas at merali dot org>
patch by: Sebastian Pölsterl

60
gst/rtpmanager/gstrtpbin.c
View file

@ -120,6 +120,7 @@
#include "gstrtpbin-marshal.h"
#include "gstrtpbin.h"
#include "gstrtpsession.h"
#include "gstrtpjitterbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
#define GST_CAT_DEFAULT gst_rtp_bin_debug
@ -236,6 +237,7 @@ enum
SIGNAL_ON_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
SIGNAL_ON_SENDER_TIMEOUT,
LAST_SIGNAL
};
@ -323,7 +325,6 @@ struct _GstRtpBinStream
guint64 clock_base_time;
gint clock_rate;
gint64 ts_offset;
gint64 prev_ts_offset;
gint last_pt;
};
@ -455,6 +456,13 @@ on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
sess->id, ssrc);
}
static void
on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
{
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
sess->id, ssrc);
}
/* create a session with the given id. Must be called with RTP_BIN_LOCK */
static GstRtpBinSession *
create_session (GstRtpBin * rtpbin, gint id)
@ -507,6 +515,8 @@ create_session (GstRtpBin * rtpbin, gint id)
g_signal_connect (sess->session, "on-bye-timeout",
(GCallback) on_bye_timeout, sess);
g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
g_signal_connect (sess->session, "on-sender-timeout",
(GCallback) on_sender_timeout, sess);
/* FIXME, change state only to what's needed */
gst_bin_add (GST_BIN_CAST (rtpbin), session);
@ -863,32 +873,31 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
/* calculate offsets for each stream */
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
if (ostream->unix_delta == 0)
continue;
gint64 prev_ts_offset;
ostream->ts_offset = ostream->unix_delta - min;
g_object_get (ostream->buffer, "ts-offset", &prev_ts_offset, NULL);
/* delta changed, see how much */
if (ostream->prev_ts_offset != ostream->ts_offset) {
if (prev_ts_offset != ostream->ts_offset) {
gint64 diff;
if (ostream->prev_ts_offset > ostream->ts_offset)
diff = ostream->prev_ts_offset - ostream->ts_offset;
if (prev_ts_offset > ostream->ts_offset)
diff = prev_ts_offset - ostream->ts_offset;
else
diff = ostream->ts_offset - ostream->prev_ts_offset;
diff = ostream->ts_offset - prev_ts_offset;
GST_DEBUG_OBJECT (bin,
"ts-offset %" G_GUINT64_FORMAT ", prev %" G_GUINT64_FORMAT
", diff: %" G_GINT64_FORMAT, ostream->ts_offset,
ostream->prev_ts_offset, diff);
", diff: %" G_GINT64_FORMAT, ostream->ts_offset, prev_ts_offset,
diff);
/* only change diff when it changed more than 1 millisecond. This
/* only change diff when it changed more than 4 milliseconds. This
* compensates for rounding errors in NTP to RTP timestamp
* conversions */
if (diff > GST_MSECOND && diff < (3 * GST_SECOND)) {
if (diff > 4 * GST_MSECOND && diff < (3 * GST_SECOND)) {
g_object_set (ostream->buffer, "ts-offset", ostream->ts_offset, NULL);
ostream->prev_ts_offset = ostream->ts_offset;
}
}
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
@ -937,8 +946,7 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
gboolean have_sr, have_sdes;
gboolean more;
guint64 clock_base;
clock_base = GST_BUFFER_OFFSET (buffer);
guint64 clock_base_time;
stream = gst_pad_get_element_private (pad);
bin = stream->bin;
@ -948,6 +956,12 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
if (!gst_rtcp_buffer_validate (buffer))
goto invalid_rtcp;
/* get the last relation between the rtp timestamps and the gstreamer
* timestamps. We get this info directly from the jitterbuffer which
* constructs gstreamer timestamps from rtp timestamps */
gst_rtp_jitter_buffer_get_sync (GST_RTP_JITTER_BUFFER (stream->buffer),
&clock_base, &clock_base_time);
/* clock base changes when there is a huge gap in the timestamps or seqnum.
* When this happens we don't want to calculate the extended timestamp based
* on the previous one but reset the calculation. */
@ -1008,7 +1022,7 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
if (type == GST_RTCP_SDES_CNAME) {
stream->clock_base = clock_base;
stream->clock_base_time = GST_BUFFER_OFFSET_END (buffer);
stream->clock_base_time = clock_base_time;
/* associate the stream to CNAME */
gst_rtp_bin_associate (bin, stream, len, data);
}
@ -1328,6 +1342,19 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::on-sender-timeout:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
*
* Notify of a sender SSRC that has timed out and became a receiver
*/
gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
G_TYPE_UINT, G_TYPE_UINT);
g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
g_param_spec_string ("sdes-cname", "SDES CNAME",
@ -2332,6 +2359,7 @@ gst_rtp_bin_request_new_pad (GstElement * element,
GstRtpBin *rtpbin;
GstElementClass *klass;
GstPad *result;
gchar *pad_name = NULL;
g_return_val_if_fail (templ != NULL, NULL);

1
gst/rtpmanager/gstrtpbin.h
View file

@ -74,6 +74,7 @@ struct _GstRtpBinClass {
void (*on_bye_ssrc) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
void (*on_bye_timeout) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
void (*on_timeout) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
void (*on_sender_timeout) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
};
GType gst_rtp_bin_get_type (void);

80
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -65,6 +65,7 @@
#include "gstrtpjitterbuffer.h"
#include "rtpjitterbuffer.h"
#include "rtpstats.h"
GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
#define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
@ -108,7 +109,7 @@ enum
#define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
JBUF_LOCK (priv); \
if (priv->srcresult != GST_FLOW_OK) \
if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
goto label; \
} G_STMT_END
@ -117,7 +118,7 @@ enum
#define JBUF_WAIT_CHECK(priv,label) G_STMT_START { \
JBUF_WAIT(priv); \
if (priv->srcresult != GST_FLOW_OK) \
if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
goto label; \
} G_STMT_END
@ -830,12 +831,12 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (!gst_rtp_buffer_validate (buffer))
if (G_UNLIKELY (!gst_rtp_buffer_validate (buffer)))
goto invalid_buffer;
priv = jitterbuffer->priv;
if (priv->last_pt != gst_rtp_buffer_get_payload_type (buffer)) {
if (G_UNLIKELY (priv->last_pt != gst_rtp_buffer_get_payload_type (buffer))) {
GstCaps *caps;
priv->last_pt = gst_rtp_buffer_get_payload_type (buffer);
@ -848,14 +849,14 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
}
}
if (priv->clock_rate == -1) {
if (G_UNLIKELY (priv->clock_rate == -1)) {
guint8 pt;
/* no clock rate given on the caps, try to get one with the signal */
pt = gst_rtp_buffer_get_payload_type (buffer);
gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
if (priv->clock_rate == -1)
if (G_UNLIKELY (priv->clock_rate == -1))
goto not_negotiated;
}
@ -875,35 +876,42 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
JBUF_LOCK_CHECK (priv, out_flushing);
/* don't accept more data on EOS */
if (priv->eos)
if (G_UNLIKELY (priv->eos))
goto have_eos;
/* let's check if this buffer is too late, we can only accept packets with
* bigger seqnum than the one we last pushed. */
if (priv->last_popped_seqnum != -1) {
if (G_LIKELY (priv->last_popped_seqnum != -1)) {
gint gap;
gboolean reset = FALSE;
gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
if (gap <= 0) {
if (G_UNLIKELY (gap <= 0)) {
/* priv->last_popped_seqnum >= seqnum, this packet is too late or the
* sender might have been restarted with different seqnum. */
if (gap < -100) {
if (gap < -RTP_MAX_MISORDER) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
reset = TRUE;
} else {
goto too_late;
}
} else {
/* priv->last_popped_seqnum < seqnum, this is a new packet */
if (gap > 3000) {
if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
reset = TRUE;
} else {
GST_DEBUG_OBJECT (jitterbuffer, "dropped packets %d but <= %d", gap,
RTP_MAX_DROPOUT);
}
}
if (G_UNLIKELY (reset)) {
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
rtp_jitter_buffer_reset_skew (priv->jbuf);
}
}
/* let's drop oldest packet if the queue is already full and drop-on-latency
@ -915,7 +923,7 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
latency_ts =
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
if (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts) {
if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
GstBuffer *old_buf;
old_buf = rtp_jitter_buffer_pop (priv->jbuf);
@ -934,8 +942,8 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
/* now insert the packet into the queue in sorted order. This function returns
* FALSE if a packet with the same seqnum was already in the queue, meaning we
* have a duplicate. */
if (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
priv->clock_rate, &tail))
if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
priv->clock_rate, &tail)))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */
@ -944,7 +952,7 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
/* let's unschedule and unblock any waiting buffers. We only want to do this
* when the tail buffer changed */
if (priv->clock_id && tail) {
if (G_UNLIKELY (priv->clock_id && tail)) {
GST_DEBUG_OBJECT (jitterbuffer,
"Unscheduling waiting buffer, new tail buffer");
gst_clock_id_unschedule (priv->clock_id);
@ -1051,12 +1059,12 @@ again:
GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
while (TRUE) {
/* always wait if we are blocked */
if (!priv->blocked) {
if (G_LIKELY (!priv->blocked)) {
/* if we have a packet, we can exit the loop and grab it */
if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
break;
/* no packets but we are EOS, do eos logic */
if (priv->eos)
if (G_UNLIKELY (priv->eos))
goto do_eos;
}
/* underrun, wait for packets or flushing now */
@ -1091,12 +1099,12 @@ again:
/* get the gap between this and the previous packet. If we don't know the
* previous packet seqnum assume no gap. */
if (next_seqnum != -1) {
if (G_LIKELY (next_seqnum != -1)) {
gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
/* if we have a packet that we already pushed or considered dropped, pop it
* off and get the next packet */
if (gap < 0) {
if (G_UNLIKELY (gap < 0)) {
GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
seqnum, next_seqnum);
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
@ -1116,7 +1124,7 @@ again:
* determine if we have missing a packet. If we have a missing packet (which
* must be before this packet) we can wait for it until the deadline for this
* packet expires. */
if (gap != 0 && out_time != -1) {
if (G_UNLIKELY (gap != 0 && out_time != -1)) {
GstClockID id;
GstClockTime sync_time;
GstClockReturn ret;
@ -1188,8 +1196,9 @@ again:
/* at this point, the clock could have been unlocked by a timeout, a new
* tail element was added to the queue or because we are shutting down. Check
* for shutdown first. */
if (priv->srcresult != GST_FLOW_OK)
goto flushing;
if G_UNLIKELY
((priv->srcresult != GST_FLOW_OK))
goto flushing;
/* if we got unscheduled and we are not flushing, it's because a new tail
* element became available in the queue. Grab it and try to push or sync. */
@ -1239,7 +1248,7 @@ push_buffer:
/* when we get here we are ready to pop and push the buffer */
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
if (discont || priv->discont) {
if (G_UNLIKELY (discont || priv->discont)) {
/* set DISCONT flag when we missed a packet. We pushed the buffer writable
* into the jitterbuffer so we can modify now. */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
@ -1261,7 +1270,7 @@ push_buffer:
"Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (out_time));
result = gst_pad_push (priv->srcpad, outbuf);
if (result != GST_FLOW_OK)
if (G_UNLIKELY (result != GST_FLOW_OK))
goto pause;
return;
@ -1451,3 +1460,18 @@ gst_rtp_jitter_buffer_get_property (GObject * object,
break;
}
}
void
gst_rtp_jitter_buffer_get_sync (GstRtpJitterBuffer * buffer, guint64 * rtptime,
guint64 * timestamp)
{
GstRtpJitterBufferPrivate *priv;
g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (buffer));
priv = buffer->priv;
JBUF_LOCK (priv);
rtp_jitter_buffer_get_sync (priv->jbuf, rtptime, timestamp);
JBUF_UNLOCK (priv);
}

3
gst/rtpmanager/gstrtpjitterbuffer.h
View file

@ -79,6 +79,9 @@ struct _GstRtpJitterBufferClass
GType gst_rtp_jitter_buffer_get_type (void);
void gst_rtp_jitter_buffer_get_sync (GstRtpJitterBuffer *buffer,
guint64 *rtptime, guint64 *timestamp);
G_END_DECLS
#endif /* __GST_RTP_JITTER_BUFFER_H__ */

23
gst/rtpmanager/gstrtpsession.c
View file

@ -193,6 +193,7 @@ enum
SIGNAL_ON_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
SIGNAL_ON_SENDER_TIMEOUT,
LAST_SIGNAL
};
@ -416,6 +417,13 @@ on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
src->ssrc);
}
static void
on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
src->ssrc);
}
GST_BOILERPLATE (GstRtpSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
static void
@ -574,6 +582,18 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_timeout),
NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-sender-timeout:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of a sender SSRC that has timed out and became a receiver
*/
gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT,
G_TYPE_NONE, 1, G_TYPE_UINT);
g_object_class_install_property (gobject_class, PROP_NTP_NS_BASE,
g_param_spec_uint64 ("ntp-ns-base", "NTP base time",
@ -655,6 +675,7 @@ gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass)
rtpsession->priv->lock = g_mutex_new ();
rtpsession->priv->sysclock = gst_system_clock_obtain ();
rtpsession->priv->session = rtp_session_new ();
/* configure callbacks */
rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
/* configure signals */
@ -674,6 +695,8 @@ gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass)
(GCallback) on_bye_timeout, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-timeout",
(GCallback) on_timeout, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
(GCallback) on_sender_timeout, rtpsession);
rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);

1
gst/rtpmanager/gstrtpsession.h
View file

@ -71,6 +71,7 @@ struct _GstRtpSessionClass {
void (*on_bye_ssrc) (GstRtpSession *sess, guint32 ssrc);
void (*on_bye_timeout) (GstRtpSession *sess, guint32 ssrc);
void (*on_timeout) (GstRtpSession *sess, guint32 ssrc);
void (*on_sender_timeout) (GstRtpSession *sess, guint32 ssrc);
};
GType gst_rtp_session_get_type (void);

82
gst/rtpmanager/rtpjitterbuffer.c
View file

@ -104,6 +104,7 @@ rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
{
jbuf->base_time = -1;
jbuf->base_rtptime = -1;
jbuf->base_extrtp = -1;
jbuf->ext_rtptime = -1;
jbuf->window_pos = 0;
jbuf->window_filling = TRUE;
@ -185,21 +186,23 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
gstrtptime = gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, clock_rate);
again:
/* first time, lock on to time and gstrtptime */
if (jbuf->base_time == -1)
if (G_UNLIKELY (jbuf->base_time == -1))
jbuf->base_time = time;
if (jbuf->base_rtptime == -1)
if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
jbuf->base_rtptime = gstrtptime;
jbuf->base_extrtp = ext_rtptime;
}
if (gstrtptime >= jbuf->base_rtptime)
if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
send_diff = gstrtptime - jbuf->base_rtptime;
else {
/* elapsed time at sender, timestamps can go backwards and thus be smaller
* than our base time, take a new base time in that case. */
GST_DEBUG ("backward timestamps at server, taking new base time");
jbuf->base_rtptime = gstrtptime;
jbuf->base_time = time;
jbuf->base_rtptime = gstrtptime;
jbuf->base_extrtp = ext_rtptime;
send_diff = 0;
}
@ -208,27 +211,6 @@ again:
GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
GST_TIME_ARGS (send_diff));
if (jbuf->prev_send_diff != -1 && time != -1) {
gint64 delta_diff;
if (send_diff > jbuf->prev_send_diff)
delta_diff = send_diff - jbuf->prev_send_diff;
else
delta_diff = jbuf->prev_send_diff - send_diff;
/* server changed rtp timestamps too quickly, reset skew detection and start
* again. This value is sortof arbitrary and can be a bad measurement up if
* there are many packets missing because then we get a big gap that is
* unrelated to a timestamp switch. */
if (delta_diff > GST_SECOND) {
GST_DEBUG ("delta changed too quickly %" GST_TIME_FORMAT " reset skew",
GST_TIME_ARGS (delta_diff));
rtp_jitter_buffer_reset_skew (jbuf);
goto again;
}
}
jbuf->prev_send_diff = send_diff;
/* we don't have an arrival timestamp so we can't do skew detection. we
* should still apply a timestamp based on RTP timestamp and base_time */
if (time == -1)
@ -244,17 +226,30 @@ again:
/* measure the diff */
delta = ((gint64) recv_diff) - ((gint64) send_diff);
/* if the difference between the sender timeline and the receiver timeline
* changed too quickly we have to resync because the server likely restarted
* its timestamps. */
if (ABS (delta - jbuf->skew) > GST_SECOND) {
GST_DEBUG ("delta %" GST_TIME_FORMAT " too big, reset skew",
delta - jbuf->skew);
jbuf->base_time = time;
jbuf->base_rtptime = gstrtptime;
jbuf->base_extrtp = ext_rtptime;
send_diff = 0;
delta = 0;
}
pos = jbuf->window_pos;
if (jbuf->window_filling) {
if (G_UNLIKELY (jbuf->window_filling)) {
/* we are filling the window */
GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
jbuf->window[pos++] = delta;
/* calc the min delta we observed */
if (pos == 1 || delta < jbuf->window_min)
if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
jbuf->window_min = delta;
if (send_diff >= MAX_TIME || pos >= MAX_WINDOW) {
if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
jbuf->window_size = pos;
/* window filled */
@ -288,11 +283,11 @@ again:
old = jbuf->window[pos];
jbuf->window[pos++] = delta;
if (delta <= jbuf->window_min) {
if (G_UNLIKELY (delta <= jbuf->window_min)) {
/* if the new value we inserted is smaller or equal to the current min,
* it becomes the new min */
jbuf->window_min = delta;
} else if (old == jbuf->window_min) {
} else if (G_UNLIKELY (old == jbuf->window_min)) {
gint64 min = G_MAXINT64;
/* if we removed the old min, we have to find a new min */
@ -313,7 +308,7 @@ again:
delta, jbuf->window_min);
}
/* wrap around in the window */
if (pos >= jbuf->window_size)
if (G_UNLIKELY (pos >= jbuf->window_size))
pos = 0;
jbuf->window_pos = pos;
@ -382,14 +377,14 @@ rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
time = calculate_skew (jbuf, rtptime, time, clock_rate);
GST_BUFFER_TIMESTAMP (buf) = time;
if (list)
if (G_LIKELY (list))
g_queue_insert_before (jbuf->packets, list, buf);
else
g_queue_push_tail (jbuf->packets, buf);
/* tail was changed when we did not find a previous packet, we set the return
* flag when requested. */
if (tail)
if (G_UNLIKELY (tail))
*tail = (list == NULL);
return TRUE;
@ -514,3 +509,22 @@ rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
}
return result;
}
/**
* rtp_jitter_buffer_get_sync:
* @jbuf: an #RTPJitterBuffer
* @rtptime: result RTP time
* @timestamp: result GStreamer timestamp
*
* Returns the relation between the RTP timestamp and the GStreamer timestamp
* used for constructing timestamps.
*/
void
rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
guint64 * timestamp)
{
if (rtptime)
*rtptime = jbuf->base_extrtp;
if (timestamp)
*timestamp = jbuf->base_time + jbuf->skew;
}

6
gst/rtpmanager/rtpjitterbuffer.h
View file

@ -22,7 +22,6 @@
#include <gst/gst.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/netbuffer/gstnetbuffer.h>
typedef struct _RTPJitterBuffer RTPJitterBuffer;
typedef struct _RTPJitterBufferClass RTPJitterBufferClass;
@ -57,6 +56,7 @@ struct _RTPJitterBuffer {
/* for calculating skew */
GstClockTime base_time;
GstClockTime base_rtptime;
GstClockTime base_extrtp;
guint64 ext_rtptime;
gint64 window[RTP_JITTER_BUFFER_MAX_WINDOW];
guint window_pos;
@ -90,4 +90,8 @@ void rtp_jitter_buffer_flush (RTPJitterBuffer *jbuf)
guint rtp_jitter_buffer_num_packets (RTPJitterBuffer *jbuf);
guint32 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer *jbuf);
void rtp_jitter_buffer_get_sync (RTPJitterBuffer *jbuf, guint64 *rtptime,
guint64 *timestamp);
#endif /* __RTP_JITTER_BUFFER_H__ */

36
gst/rtpmanager/rtpsession.c
View file

@ -40,6 +40,7 @@ enum
SIGNAL_ON_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
SIGNAL_ON_SENDER_TIMEOUT,
LAST_SIGNAL
};
@ -212,6 +213,18 @@ rtp_session_class_init (RTPSessionClass * klass)
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-sender-timeout:
* @session: the object which received the signal
* @src: the RTPSource that timed out
*
* Notify of an SSRC that was a sender but timed out and became a receiver.
*/
rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
g_param_spec_object ("internal-source", "Internal Source",
@ -513,6 +526,15 @@ on_timeout (RTPSession * sess, RTPSource * source)
RTP_SESSION_LOCK (sess);
}
static void
on_sender_timeout (RTPSession * sess, RTPSource * source)
{
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
source);
RTP_SESSION_LOCK (sess);
}
/**
* rtp_session_new:
*
@ -908,9 +930,8 @@ check_collision (RTPSession * sess, RTPSource * source,
RTPArrivalStats * arrival, gboolean rtp)
{
/* If we have not arrival address, we can't do collision checking */
if (!arrival->have_address) {
if (!arrival->have_address)
return FALSE;
}
if (sess->source != source) {
/* This is not our local source, but lets check if two remote
@ -1479,12 +1500,6 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
if (!source)
return;
/* we somehow need to transfer the clock_base and the base time to the next
* element, we use the offset and offset_end fields in the buffer for this
* hack */
GST_BUFFER_OFFSET (packet->buffer) = source->clock_base;
GST_BUFFER_OFFSET_END (packet->buffer) = source->clock_base_time;
prevsender = RTP_SOURCE_IS_SENDER (source);
/* first update the source */
@ -2096,6 +2111,7 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
{
gboolean remove = FALSE;
gboolean byetimeout = FALSE;
gboolean sendertimeout = FALSE;
gboolean is_sender, is_active;
RTPSession *sess = data->sess;
GstClockTime interval;
@ -2138,6 +2154,7 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
GST_TIME_ARGS (source->last_rtp_activity));
source->is_sender = FALSE;
sess->stats.sender_sources--;
sendertimeout = TRUE;
}
}
}
@ -2153,6 +2170,9 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
on_bye_timeout (sess, source);
else
on_timeout (sess, source);
} else {
if (sendertimeout)
on_sender_timeout (sess, source);
}
return remove;
}

1
gst/rtpmanager/rtpsession.h
View file

@ -228,6 +228,7 @@ struct _RTPSessionClass {
void (*on_bye_ssrc) (RTPSession *sess, RTPSource *source);
void (*on_bye_timeout) (RTPSession *sess, RTPSource *source);
void (*on_timeout) (RTPSession *sess, RTPSource *source);
void (*on_sender_timeout) (RTPSession *sess, RTPSource *source);
};
GType rtp_session_get_type (void);

15
gst/rtpmanager/rtpsource.c
View file

@ -170,8 +170,6 @@ rtp_source_init (RTPSource * src)
src->payload = 0;
src->clock_rate = -1;
src->clock_base = -1;
src->clock_base_time = -1;
src->packets = g_queue_new ();
src->seqnum_base = -1;
src->last_rtptime = -1;
@ -527,10 +525,6 @@ rtp_source_update_caps (RTPSource * src, GstCaps * caps)
gst_structure_get_int (s, "clock-rate", &src->clock_rate);
GST_DEBUG ("got clock-rate %d", src->clock_rate);
if (gst_structure_get_uint (s, "clock-base", &val))
src->clock_base = val;
GST_DEBUG ("got clock-base %" G_GINT64_FORMAT, src->clock_base);
if (gst_structure_get_uint (s, "seqnum-base", &val))
src->seqnum_base = val;
GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
@ -771,13 +765,6 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
rtptime = gst_rtp_buffer_get_timestamp (buffer);
/* no clock-base, take first rtptime as base */
if (src->clock_base == -1) {
GST_DEBUG ("using clock-base of %" G_GUINT32_FORMAT, rtptime);
src->clock_base = rtptime;
src->clock_base_time = arrival->timestamp;
}
/* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
* care about the absolute value, just the difference. */
rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND);
@ -923,13 +910,11 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
} else {
/* unacceptable jump */
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
src->clock_base = -1;
goto bad_sequence;
}
} else {
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
GST_WARNING ("duplicate or reordered packet");
src->clock_base = -1;
}
src->stats.octets_received += arrival->payload_len;

4
gst/rtpmanager/rtpsource.h
View file

@ -32,8 +32,6 @@
#define RTP_DEFAULT_PROBATION 2
#define RTP_SEQ_MOD (1 << 16)
#define RTP_MAX_DROPOUT 3000
#define RTP_MAX_MISORDER 100
typedef struct _RTPSource RTPSource;
typedef struct _RTPSourceClass RTPSourceClass;
@ -133,8 +131,6 @@ struct _RTPSource {
GstCaps *caps;
gint clock_rate;
gint32 seqnum_base;
gint64 clock_base;
guint64 clock_base_time;
GstClockTime bye_time;
GstClockTime last_activity;

13
gst/rtpmanager/rtpstats.h
View file

@ -150,11 +150,22 @@ typedef struct {
#define RTP_STATS_RECEIVER_FRACTION (1.0 - RTP_STATS_SENDER_FRACTION)
/*
* When receiving a BYE from a source, remove the source fomr the database
* When receiving a BYE from a source, remove the source from the database
* after this timeout.
*/
#define RTP_STATS_BYE_TIMEOUT (2 * GST_SECOND)
/*
* The maximum number of missing packets we tollerate. These are packets with a
* sequence number bigger than the last seen packet.
*/
#define RTP_MAX_DROPOUT 3000
/*
* The maximum number of misordered packets we tollerate. These are packets with
* a sequence number smaller than the last seen packet.
*/
#define RTP_MAX_MISORDER 100
/**
* RTPSessionStats:
*