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rtpbin: alternative inter-stream syncing methods

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... at least if not syncing to NPT time:
* either sync using RTCP SR data (as currently)
* only perform the above once using initial RTCP SR packets
* discard RTCP and sync by equating provided stream's clock-base rtptime,
  as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).
This commit is contained in:
Mark Nauwelaerts 2011-08-08 12:15:20 +02:00
parent 4b7301e4d1
commit 9c95072048
2 changed files with 161 additions and 11 deletions
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169
gst/rtpmanager/gstrtpbin.c
View file

@ -248,6 +248,7 @@ enum
#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
enum
@ -258,6 +259,7 @@ enum
PROP_DO_LOST,
PROP_IGNORE_PT,
PROP_NTP_SYNC,
PROP_RTCP_SYNC,
PROP_RTCP_SYNC_INTERVAL,
PROP_AUTOREMOVE,
PROP_BUFFER_MODE,
@ -265,6 +267,31 @@ enum
PROP_LAST
};
enum
{
GST_RTP_BIN_RTCP_SYNC_ALWAYS,
GST_RTP_BIN_RTCP_SYNC_INITIAL,
GST_RTP_BIN_RTCP_SYNC_RTP
};
#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
static GType
gst_rtp_bin_rtcp_sync_get_type (void)
{
static GType rtcp_sync_type = 0;
static const GEnumValue rtcp_sync_types[] = {
{GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
{GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
{GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
{0, NULL, NULL},
};
if (!rtcp_sync_type) {
rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
}
return rtcp_sync_type;
}
/* helper objects */
typedef struct _GstRtpBinSession GstRtpBinSession;
typedef struct _GstRtpBinStream GstRtpBinStream;
@ -315,6 +342,9 @@ struct _GstRtpBinStream
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)
@ -780,6 +810,8 @@ gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
* 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);
@ -984,7 +1016,8 @@ stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
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)
guint64 base_rtptime, guint64 base_time, guint clock_rate,
gint64 rtp_clock_base)
{
GstRtpBinClient *client;
gboolean created;
@ -1027,8 +1060,9 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
GST_DEBUG_OBJECT (bin,
"base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", base_rtptime,
last_extrtptime, local_rtp, clock_rate);
", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
"clock-base %" G_GINT64_FORMAT, base_rtptime,
last_extrtptime, local_rtp, clock_rate, rtp_clock_base);
/* calculate local RTP time in gstreamer timestamp, we essentially perform the
* same conversion that a jitterbuffer would use to convert an rtp timestamp
@ -1075,8 +1109,10 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
stream->rt_delta = rtdiff - ntpdiff;
stream_set_ts_offset (bin, stream, stream->rt_delta);
} else if (client->nstreams > 1) {
gint64 min;
} else {
gint64 min, rtp_min, clock_base = stream->clock_base;
gboolean all_sync, use_rtp;
gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
/* calculate delta between server and receiver. last_unix is created by
* converting the ntptime in the last SR packet to a gstreamer timestamp. This
@ -1094,19 +1130,104 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
* 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. */
min = G_MAXINT64;
/* 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;
}
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
if (!ostream->have_sync)
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;
}
GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT, client,
min);
/* arrange to re-sync for each stream upon significant change,
* e.g. post-seek */
all_sync = (stream->clock_base == clock_base);
stream->clock_base = clock_base;
/* may need init performed above later on, but nothing more to do now */
if (client->nstreams <= 1)
return;
GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
" all sync %d", client, min, all_sync);
GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
switch (rtcp_sync) {
case GST_RTP_BIN_RTCP_SYNC_RTP:
if (!use_rtp)
break;
GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
"client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
/* fall-through */
case GST_RTP_BIN_RTCP_SYNC_INITIAL:
/* if all have been synced already, do not bother further */
if (all_sync) {
GST_DEBUG_OBJECT (bin, "all streams already synced; done");
return;
}
break;
default:
break;
}
/* bail out if we adjusted recently enough */
if (all_sync && (last_unix - bin->priv->last_unix) <
@ -1131,6 +1252,9 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
/* 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);
@ -1164,6 +1288,7 @@ gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
guint64 base_rtptime;
guint64 base_time;
guint clock_rate;
guint64 clock_base;
guint64 extrtptime;
GstBuffer *buffer;
@ -1179,6 +1304,7 @@ gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
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"));
@ -1231,7 +1357,8 @@ gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
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);
ntptime, extrtptime, base_rtptime, base_time, clock_rate,
clock_base);
GST_RTP_BIN_UNLOCK (bin);
}
}
@ -1275,7 +1402,9 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
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 */
@ -1692,6 +1821,19 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
"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.
*
* Since: 0.10.31
*/
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:
*
@ -1734,6 +1876,7 @@ gst_rtp_bin_init (GstRtpBin * rtpbin, GstRtpBinClass * klass)
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;
@ -1850,6 +1993,9 @@ gst_rtp_bin_set_property (GObject * object, guint prop_id,
case PROP_NTP_SYNC:
rtpbin->ntp_sync = g_value_get_boolean (value);
break;
case PROP_RTCP_SYNC:
g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
break;
case PROP_RTCP_SYNC_INTERVAL:
rtpbin->rtcp_sync_interval = g_value_get_uint (value);
break;
@ -1915,6 +2061,9 @@ gst_rtp_bin_get_property (GObject * object, guint prop_id,
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;

1
gst/rtpmanager/gstrtpbin.h
View file

@ -50,6 +50,7 @@ struct _GstRtpBin {
gboolean do_lost;
gboolean ignore_pt;
gboolean ntp_sync;
gint rtcp_sync;
guint rtcp_sync_interval;
RTPJitterBufferMode buffer_mode;
gboolean buffering;