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gstreamer/gst/rtpmanager/rtpsource.c
Nicolas Dufresne 6bb53e75fb rtpsession: Send as many nack seqnum as possible 
In order to do that, we now split the nacks registration from the actual
FB nack packet construction. We then try and add as many FB Nacks as
possible into the active packets and leave the remaining seqnums in the
RTPSource. In order to avoid sending outdated NACK later on, we save the
seqnum calculated deadline and cleanup the outdated seqnums before the
next RTCP send.

Fixes #583
2019-04-05 14:53:09 +00:00

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
* Copyright (C) 2015 Kurento (http://kurento.org/)
* @author: Miguel París <mparisdiaz@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.
*/
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "rtpsource.h"
GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
#define GST_CAT_DEFAULT rtp_source_debug
#define RTP_MAX_PROBATION_LEN 32
/* signals and args */
enum
{
LAST_SIGNAL
};
#define DEFAULT_SSRC 0
#define DEFAULT_IS_CSRC FALSE
#define DEFAULT_IS_VALIDATED FALSE
#define DEFAULT_IS_SENDER FALSE
#define DEFAULT_SDES NULL
#define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
#define DEFAULT_MAX_DROPOUT_TIME 60000
#define DEFAULT_MAX_MISORDER_TIME 2000
#define DEFAULT_DISABLE_RTCP FALSE
enum
{
PROP_0,
PROP_SSRC,
PROP_IS_CSRC,
PROP_IS_VALIDATED,
PROP_IS_SENDER,
PROP_SDES,
PROP_STATS,
PROP_PROBATION,
PROP_MAX_DROPOUT_TIME,
PROP_MAX_MISORDER_TIME,
PROP_DISABLE_RTCP
};
/* GObject vmethods */
static void rtp_source_finalize (GObject * object);
static void rtp_source_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void rtp_source_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
static void
rtp_source_class_init (RTPSourceClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = rtp_source_finalize;
gobject_class->set_property = rtp_source_set_property;
gobject_class->get_property = rtp_source_get_property;
g_object_class_install_property (gobject_class, PROP_SSRC,
g_param_spec_uint ("ssrc", "SSRC",
"The SSRC of this source", 0, G_MAXUINT, DEFAULT_SSRC,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IS_CSRC,
g_param_spec_boolean ("is-csrc", "Is CSRC",
"If this SSRC is acting as a contributing source",
DEFAULT_IS_CSRC, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IS_VALIDATED,
g_param_spec_boolean ("is-validated", "Is Validated",
"If this SSRC is validated", DEFAULT_IS_VALIDATED,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IS_SENDER,
g_param_spec_boolean ("is-sender", "Is Sender",
"If this SSRC is a sender", DEFAULT_IS_SENDER,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* RTPSource::sdes
*
* The current SDES items of the source. Returns a structure with name
* application/x-rtp-source-sdes and may contain the following fields:
*
* 'cname' G_TYPE_STRING : The canonical name in the form user@host
* 'name' G_TYPE_STRING : The user name
* 'email' G_TYPE_STRING : The user's electronic mail address
* 'phone' G_TYPE_STRING : The user's phone number
* 'location' G_TYPE_STRING : The geographic user location
* 'tool' G_TYPE_STRING : The name of application or tool
* 'note' G_TYPE_STRING : A notice about the source
*
* Other fields may be present and these represent private items in
* the SDES where the field name is the prefix.
*/
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES information for this source",
GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* RTPSource::stats
*
* This property returns a GstStructure named application/x-rtp-source-stats with
* fields useful for statistics and diagnostics.
*
* Take note of each respective field's units:
*
* - NTP times are in the appropriate 32-bit or 64-bit fixed-point format
* starting from January 1, 1970 (except for timespans).
* - RTP times are in clock rate units (i.e. clock rate = 1 second)
* starting at a random offset.
* - For fields indicating packet loss, note that late packets are not considered lost,
* and duplicates are not taken into account. Hence, the loss may be negative
* if there are duplicates.
*
* The following fields are always present.
*
* "ssrc" G_TYPE_UINT the SSRC of this source
* "internal" G_TYPE_BOOLEAN this source is a source of the session
* "validated" G_TYPE_BOOLEAN the source is validated
* "received-bye" G_TYPE_BOOLEAN we received a BYE from this source
* "is-csrc" G_TYPE_BOOLEAN this source was found as CSRC
* "is-sender" G_TYPE_BOOLEAN this source is a sender
* "seqnum-base" G_TYPE_INT first seqnum if known
* "clock-rate" G_TYPE_INT the clock rate of the media
*
* The following fields are only present when known.
*
* "rtp-from" G_TYPE_STRING where we received the last RTP packet from
* "rtcp-from" G_TYPE_STRING where we received the last RTCP packet from
*
* The following fields make sense for internal sources and will only increase
* when "is-sender" is TRUE.
*
* "octets-sent" G_TYPE_UINT64 number of bytes we sent
* "packets-sent" G_TYPE_UINT64 number of packets we sent
*
* The following fields make sense for non-internal sources and will only
* increase when "is-sender" is TRUE.
*
* "octets-received" G_TYPE_UINT64 total number of bytes received
* "packets-received" G_TYPE_UINT64 total number of packets received
*
* Following fields are updated when "is-sender" is TRUE.
*
* "bitrate" G_TYPE_UINT64 bitrate in bits per second
* "jitter" G_TYPE_UINT estimated jitter (in clock rate units)
* "packets-lost" G_TYPE_INT estimated amount of packets lost
*
* The last SR report this source sent. This only updates when "is-sender" is
* TRUE.
*
* "have-sr" G_TYPE_BOOLEAN the source has sent SR
* "sr-ntptime" G_TYPE_UINT64 NTP time of SR (in NTP Timestamp Format, 32.32 fixed point)
* "sr-rtptime" G_TYPE_UINT RTP time of SR (in clock rate units)
* "sr-octet-count" G_TYPE_UINT the number of bytes in the SR
* "sr-packet-count" G_TYPE_UINT the number of packets in the SR
*
* The following fields are only present for non-internal sources and
* represent the content of the last RB packet that was sent to this source.
* These values are only updated when the source is sending.
*
* "sent-rb" G_TYPE_BOOLEAN we have sent an RB
* "sent-rb-fractionlost" G_TYPE_UINT calculated lost 8-bit fraction
* "sent-rb-packetslost" G_TYPE_INT lost packets
* "sent-rb-exthighestseq" G_TYPE_UINT last seen seqnum
* "sent-rb-jitter" G_TYPE_UINT jitter (in clock rate units)
* "sent-rb-lsr" G_TYPE_UINT last SR time (seconds in NTP Short Format, 16.16 fixed point)
* "sent-rb-dlsr" G_TYPE_UINT delay since last SR (seconds in NTP Short Format, 16.16 fixed point)
*
* The following fields are only present for non-internal sources and
* represents the last RB that this source sent. This is only updated
* when the source is receiving data and sending RB blocks.
*
* "have-rb" G_TYPE_BOOLEAN the source has sent RB
* "rb-fractionlost" G_TYPE_UINT lost 8-bit fraction
* "rb-packetslost" G_TYPE_INT lost packets
* "rb-exthighestseq" G_TYPE_UINT highest received seqnum
* "rb-jitter" G_TYPE_UINT reception jitter (in clock rate units)
* "rb-lsr" G_TYPE_UINT last SR time (seconds in NTP Short Format, 16.16 fixed point)
* "rb-dlsr" G_TYPE_UINT delay since last SR (seconds in NTP Short Format, 16.16 fixed point)
*
* The round trip of this source is calculated from the last RB
* values and the reception time of the last RB packet. It is only present for
* non-internal sources.
*
* "rb-round-trip" G_TYPE_UINT the round-trip time (seconds in NTP Short Format, 16.16 fixed point)
*
*/
g_object_class_install_property (gobject_class, PROP_STATS,
g_param_spec_boxed ("stats", "Stats",
"The stats of this source", GST_TYPE_STRUCTURE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROBATION,
g_param_spec_uint ("probation", "Number of probations",
"Consecutive packet sequence numbers to accept the source",
0, G_MAXUINT, DEFAULT_PROBATION,
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));
/**
* RTPSession::disable-rtcp:
*
* Allow disabling the sending of RTCP packets for this source.
*/
g_object_class_install_property (gobject_class, PROP_DISABLE_RTCP,
g_param_spec_boolean ("disable-rtcp", "Disable RTCP",
"Disable sending RTCP packets for this source",
DEFAULT_DISABLE_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
}
/**
* rtp_source_reset:
* @src: an #RTPSource
*
* Reset the stats of @src.
*/
void
rtp_source_reset (RTPSource * src)
{
src->marked_bye = FALSE;
if (src->bye_reason)
g_free (src->bye_reason);
src->bye_reason = NULL;
src->sent_bye = FALSE;
g_hash_table_remove_all (src->reported_in_sr_of);
g_queue_foreach (src->retained_feedback, (GFunc) gst_buffer_unref, NULL);
g_queue_clear (src->retained_feedback);
src->last_rtptime = -1;
src->stats.cycles = -1;
src->stats.jitter = 0;
src->stats.transit = -1;
src->stats.curr_sr = 0;
src->stats.sr[0].is_valid = FALSE;
src->stats.curr_rr = 0;
src->stats.rr[0].is_valid = FALSE;
src->stats.prev_rtptime = GST_CLOCK_TIME_NONE;
src->stats.prev_rtcptime = GST_CLOCK_TIME_NONE;
src->stats.last_rtptime = GST_CLOCK_TIME_NONE;
src->stats.last_rtcptime = GST_CLOCK_TIME_NONE;
g_array_set_size (src->nacks, 0);
src->stats.sent_pli_count = 0;
src->stats.sent_fir_count = 0;
src->stats.sent_nack_count = 0;
src->stats.recv_nack_count = 0;
}
static void
rtp_source_init (RTPSource * src)
{
/* sources are initially on probation until we receive enough valid RTP
* packets or a valid RTCP packet */
src->validated = FALSE;
src->internal = FALSE;
src->probation = DEFAULT_PROBATION;
src->curr_probation = src->probation;
src->closing = FALSE;
src->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
src->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
src->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
src->payload = -1;
src->clock_rate = -1;
src->packets = g_queue_new ();
src->seqnum_offset = -1;
src->retained_feedback = g_queue_new ();
src->nacks = g_array_new (FALSE, FALSE, sizeof (guint16));
src->nack_deadlines = g_array_new (FALSE, FALSE, sizeof (GstClockTime));
src->reported_in_sr_of = g_hash_table_new (g_direct_hash, g_direct_equal);
src->last_keyframe_request = GST_CLOCK_TIME_NONE;
rtp_source_reset (src);
src->pt_set = FALSE;
}
void
rtp_conflicting_address_free (RTPConflictingAddress * addr)
{
g_object_unref (addr->address);
g_slice_free (RTPConflictingAddress, addr);
}
static void
rtp_source_finalize (GObject * object)
{
RTPSource *src;
src = RTP_SOURCE_CAST (object);
g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
g_queue_free (src->packets);
gst_structure_free (src->sdes);
g_free (src->bye_reason);
gst_caps_replace (&src->caps, NULL);
g_list_free_full (src->conflicting_addresses,
(GDestroyNotify) rtp_conflicting_address_free);
g_queue_foreach (src->retained_feedback, (GFunc) gst_buffer_unref, NULL);
g_queue_free (src->retained_feedback);
g_array_free (src->nacks, TRUE);
g_array_free (src->nack_deadlines, TRUE);
if (src->rtp_from)
g_object_unref (src->rtp_from);
if (src->rtcp_from)
g_object_unref (src->rtcp_from);
g_hash_table_unref (src->reported_in_sr_of);
G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
}
static GstStructure *
rtp_source_create_stats (RTPSource * src)
{
GstStructure *s;
gboolean is_sender = src->is_sender;
gboolean internal = src->internal;
gchar *address_str;
gboolean have_rb;
guint8 fractionlost = 0;
gint32 packetslost = 0;
guint32 exthighestseq = 0;
guint32 jitter = 0;
guint32 lsr = 0;
guint32 dlsr = 0;
guint32 round_trip = 0;
gboolean have_sr;
GstClockTime time = 0;
guint64 ntptime = 0;
guint32 rtptime = 0;
guint32 packet_count = 0;
guint32 octet_count = 0;
/* common data for all types of sources */
s = gst_structure_new ("application/x-rtp-source-stats",
"ssrc", G_TYPE_UINT, (guint) src->ssrc,
"internal", G_TYPE_BOOLEAN, internal,
"validated", G_TYPE_BOOLEAN, src->validated,
"received-bye", G_TYPE_BOOLEAN, src->marked_bye,
"is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
"is-sender", G_TYPE_BOOLEAN, is_sender,
"seqnum-base", G_TYPE_INT, src->seqnum_offset,
"clock-rate", G_TYPE_INT, src->clock_rate, NULL);
/* add address and port */
if (src->rtp_from) {
address_str = __g_socket_address_to_string (src->rtp_from);
gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL);
g_free (address_str);
}
if (src->rtcp_from) {
address_str = __g_socket_address_to_string (src->rtcp_from);
gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL);
g_free (address_str);
}
gst_structure_set (s,
"octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
"packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
"octets-received", G_TYPE_UINT64, src->stats.octets_received,
"packets-received", G_TYPE_UINT64, src->stats.packets_received,
"bitrate", G_TYPE_UINT64, src->bitrate,
"packets-lost", G_TYPE_INT,
(gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT,
(guint) (src->stats.jitter >> 4),
"sent-pli-count", G_TYPE_UINT, src->stats.sent_pli_count,
"recv-pli-count", G_TYPE_UINT, src->stats.recv_pli_count,
"sent-fir-count", G_TYPE_UINT, src->stats.sent_fir_count,
"recv-fir-count", G_TYPE_UINT, src->stats.recv_fir_count,
"sent-nack-count", G_TYPE_UINT, src->stats.sent_nack_count,
"recv-nack-count", G_TYPE_UINT, src->stats.recv_nack_count, NULL);
/* get the last SR. */
have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
&packet_count, &octet_count);
gst_structure_set (s,
"have-sr", G_TYPE_BOOLEAN, have_sr,
"sr-ntptime", G_TYPE_UINT64, ntptime,
"sr-rtptime", G_TYPE_UINT, (guint) rtptime,
"sr-octet-count", G_TYPE_UINT, (guint) octet_count,
"sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
if (!internal) {
/* get the last RB we sent */
gst_structure_set (s,
"sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid,
"sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost,
"sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost,
"sent-rb-exthighestseq", G_TYPE_UINT,
(guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT,
(guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT,
(guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT,
(guint) src->last_rr.dlsr, NULL);
/* get the last RB */
have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
&exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
gst_structure_set (s,
"have-rb", G_TYPE_BOOLEAN, have_rb,
"rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
"rb-packetslost", G_TYPE_INT, (gint) packetslost,
"rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
"rb-jitter", G_TYPE_UINT, (guint) jitter,
"rb-lsr", G_TYPE_UINT, (guint) lsr,
"rb-dlsr", G_TYPE_UINT, (guint) dlsr,
"rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
}
return s;
}
/**
* rtp_source_get_sdes_struct:
* @src: an #RTPSource
*
* Get the SDES from @src. See the SDES property for more details.
*
* Returns: %GstStructure of type "application/x-rtp-source-sdes". The result is
* valid until the SDES items of @src are modified.
*/
const GstStructure *
rtp_source_get_sdes_struct (RTPSource * src)
{
g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
return src->sdes;
}
static gboolean
sdes_struct_compare_func (GQuark field_id, const GValue * value,
gpointer user_data)
{
GstStructure *old;
const gchar *field;
old = GST_STRUCTURE (user_data);
field = g_quark_to_string (field_id);
if (!gst_structure_has_field (old, field))
return FALSE;
g_assert (G_VALUE_HOLDS_STRING (value));
return strcmp (g_value_get_string (value), gst_structure_get_string (old,
field)) == 0;
}
/**
* rtp_source_set_sdes_struct:
* @src: an #RTPSource
* @sdes: the SDES structure
*
* Store the @sdes in @src. @sdes must be a structure of type
* "application/x-rtp-source-sdes", see the SDES property for more details.
*
* This function takes ownership of @sdes.
*
* Returns: %FALSE if the SDES was unchanged.
*/
gboolean
rtp_source_set_sdes_struct (RTPSource * src, GstStructure * sdes)
{
gboolean changed;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
g_return_val_if_fail (strcmp (gst_structure_get_name (sdes),
"application/x-rtp-source-sdes") == 0, FALSE);
changed = !gst_structure_foreach (sdes, sdes_struct_compare_func, src->sdes);
if (changed) {
gst_structure_free (src->sdes);
src->sdes = sdes;
} else {
gst_structure_free (sdes);
}
return changed;
}
static void
rtp_source_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
RTPSource *src;
src = RTP_SOURCE (object);
switch (prop_id) {
case PROP_SSRC:
src->ssrc = g_value_get_uint (value);
break;
case PROP_PROBATION:
src->probation = g_value_get_uint (value);
break;
case PROP_MAX_DROPOUT_TIME:
src->max_dropout_time = g_value_get_uint (value);
break;
case PROP_MAX_MISORDER_TIME:
src->max_misorder_time = g_value_get_uint (value);
break;
case PROP_DISABLE_RTCP:
src->disable_rtcp = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
rtp_source_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
RTPSource *src;
src = RTP_SOURCE (object);
switch (prop_id) {
case PROP_SSRC:
g_value_set_uint (value, rtp_source_get_ssrc (src));
break;
case PROP_IS_CSRC:
g_value_set_boolean (value, rtp_source_is_as_csrc (src));
break;
case PROP_IS_VALIDATED:
g_value_set_boolean (value, rtp_source_is_validated (src));
break;
case PROP_IS_SENDER:
g_value_set_boolean (value, rtp_source_is_sender (src));
break;
case PROP_SDES:
g_value_set_boxed (value, rtp_source_get_sdes_struct (src));
break;
case PROP_STATS:
g_value_take_boxed (value, rtp_source_create_stats (src));
break;
case PROP_PROBATION:
g_value_set_uint (value, src->probation);
break;
case PROP_MAX_DROPOUT_TIME:
g_value_set_uint (value, src->max_dropout_time);
break;
case PROP_MAX_MISORDER_TIME:
g_value_set_uint (value, src->max_misorder_time);
break;
case PROP_DISABLE_RTCP:
g_value_set_boolean (value, src->disable_rtcp);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/**
* rtp_source_new:
* @ssrc: an SSRC
*
* Create a #RTPSource with @ssrc.
*
* Returns: a new #RTPSource. Use g_object_unref() after usage.
*/
RTPSource *
rtp_source_new (guint32 ssrc)
{
RTPSource *src;
src = g_object_new (RTP_TYPE_SOURCE, NULL);
src->ssrc = ssrc;
return src;
}
/**
* rtp_source_set_callbacks:
* @src: an #RTPSource
* @cb: callback functions
* @user_data: user data
*
* Set the callbacks for the source.
*/
void
rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
gpointer user_data)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->callbacks.push_rtp = cb->push_rtp;
src->callbacks.clock_rate = cb->clock_rate;
src->user_data = user_data;
}
/**
* rtp_source_get_ssrc:
* @src: an #RTPSource
*
* Get the SSRC of @source.
*
* Returns: the SSRC of src.
*/
guint32
rtp_source_get_ssrc (RTPSource * src)
{
guint32 result;
g_return_val_if_fail (RTP_IS_SOURCE (src), 0);
result = src->ssrc;
return result;
}
/**
* rtp_source_set_as_csrc:
* @src: an #RTPSource
*
* Configure @src as a CSRC, this will also validate @src.
*/
void
rtp_source_set_as_csrc (RTPSource * src)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->validated = TRUE;
src->is_csrc = TRUE;
}
/**
* rtp_source_is_as_csrc:
* @src: an #RTPSource
*
* Check if @src is a contributing source.
*
* Returns: %TRUE if @src is acting as a contributing source.
*/
gboolean
rtp_source_is_as_csrc (RTPSource * src)
{
gboolean result;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
result = src->is_csrc;
return result;
}
/**
* rtp_source_is_active:
* @src: an #RTPSource
*
* Check if @src is an active source. A source is active if it has been
* validated and has not yet received a BYE packet
*
* Returns: %TRUE if @src is an qactive source.
*/
gboolean
rtp_source_is_active (RTPSource * src)
{
gboolean result;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
result = RTP_SOURCE_IS_ACTIVE (src);
return result;
}
/**
* rtp_source_is_validated:
* @src: an #RTPSource
*
* Check if @src is a validated source.
*
* Returns: %TRUE if @src is a validated source.
*/
gboolean
rtp_source_is_validated (RTPSource * src)
{
gboolean result;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
result = src->validated;
return result;
}
/**
* rtp_source_is_sender:
* @src: an #RTPSource
*
* Check if @src is a sending source.
*
* Returns: %TRUE if @src is a sending source.
*/
gboolean
rtp_source_is_sender (RTPSource * src)
{
gboolean result;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
result = RTP_SOURCE_IS_SENDER (src);
return result;
}
/**
* rtp_source_is_marked_bye:
* @src: an #RTPSource
*
* Check if @src is marked as leaving the session with a BYE packet.
*
* Returns: %TRUE if @src has been marked BYE.
*/
gboolean
rtp_source_is_marked_bye (RTPSource * src)
{
gboolean result;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
result = RTP_SOURCE_IS_MARKED_BYE (src);
return result;
}
/**
* rtp_source_get_bye_reason:
* @src: an #RTPSource
*
* Get the BYE reason for @src. Check if the source is marked as leaving the
* session with a BYE message first with rtp_source_is_marked_bye().
*
* Returns: The BYE reason or NULL when no reason was given or the source was
* not marked BYE yet. g_free() after usage.
*/
gchar *
rtp_source_get_bye_reason (RTPSource * src)
{
gchar *result;
g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
result = g_strdup (src->bye_reason);
return result;
}
/**
* rtp_source_update_caps:
* @src: an #RTPSource
* @caps: a #GstCaps
*
* Parse @caps and store all relevant information in @source.
*/
void
rtp_source_update_caps (RTPSource * src, GstCaps * caps)
{
GstStructure *s;
guint val;
gint ival;
gboolean rtx;
/* nothing changed, return */
if (caps == NULL || src->caps == caps)
return;
s = gst_caps_get_structure (caps, 0);
rtx = (gst_structure_get_uint (s, "rtx-ssrc", &val) && val == src->ssrc);
if (gst_structure_get_int (s, rtx ? "rtx-payload" : "payload", &ival))
src->payload = ival;
else
src->payload = -1;
GST_DEBUG ("got %spayload %d", rtx ? "rtx " : "", src->payload);
if (gst_structure_get_int (s, "clock-rate", &ival))
src->clock_rate = ival;
else
src->clock_rate = -1;
GST_DEBUG ("got clock-rate %d", src->clock_rate);
if (gst_structure_get_uint (s, rtx ? "rtx-seqnum-offset" : "seqnum-offset",
&val))
src->seqnum_offset = val;
else
src->seqnum_offset = -1;
GST_DEBUG ("got %sseqnum-offset %" G_GINT32_FORMAT, rtx ? "rtx " : "",
src->seqnum_offset);
gst_caps_replace (&src->caps, caps);
}
/**
* rtp_source_set_rtp_from:
* @src: an #RTPSource
* @address: the RTP address to set
*
* Set that @src is receiving RTP packets from @address. This is used for
* collistion checking.
*/
void
rtp_source_set_rtp_from (RTPSource * src, GSocketAddress * address)
{
g_return_if_fail (RTP_IS_SOURCE (src));
if (src->rtp_from)
g_object_unref (src->rtp_from);
src->rtp_from = G_SOCKET_ADDRESS (g_object_ref (address));
}
/**
* rtp_source_set_rtcp_from:
* @src: an #RTPSource
* @address: the RTCP address to set
*
* Set that @src is receiving RTCP packets from @address. This is used for
* collistion checking.
*/
void
rtp_source_set_rtcp_from (RTPSource * src, GSocketAddress * address)
{
g_return_if_fail (RTP_IS_SOURCE (src));
if (src->rtcp_from)
g_object_unref (src->rtcp_from);
src->rtcp_from = G_SOCKET_ADDRESS (g_object_ref (address));
}
static GstFlowReturn
push_packet (RTPSource * src, GstBuffer * buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
/* push queued packets first if any */
while (!g_queue_is_empty (src->packets)) {
GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
GST_LOG ("pushing queued packet");
if (src->callbacks.push_rtp)
src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
}
GST_LOG ("pushing new packet");
/* push packet */
if (src->callbacks.push_rtp)
ret = src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
return ret;
}
static gint
get_clock_rate (RTPSource * src, guint8 payload)
{
if (src->payload == -1) {
/* first payload received, nothing was in the caps, lock on to this payload */
src->payload = payload;
GST_DEBUG ("first payload %d", payload);
} else if (payload != src->payload) {
/* we have a different payload than before, reset the clock-rate */
GST_DEBUG ("new payload %d", payload);
src->payload = payload;
src->clock_rate = -1;
src->stats.transit = -1;
}
if (src->clock_rate == -1) {
gint clock_rate = -1;
if (src->callbacks.clock_rate)
clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
GST_DEBUG ("got clock-rate %d", clock_rate);
src->clock_rate = clock_rate;
gst_rtp_packet_rate_ctx_reset (&src->packet_rate_ctx, clock_rate);
}
return src->clock_rate;
}
/* Jitter is the variation in the delay of received packets in a flow. It is
* measured by comparing the interval when RTP packets were sent to the interval
* at which they were received. For instance, if packet #1 and packet #2 leave
* 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
* milliseconds. */
static void
calculate_jitter (RTPSource * src, RTPPacketInfo * pinfo)
{
GstClockTime running_time;
guint32 rtparrival, transit, rtptime;
gint32 diff;
gint clock_rate;
guint8 pt;
/* get arrival time */
if ((running_time = pinfo->running_time) == GST_CLOCK_TIME_NONE)
goto no_time;
pt = pinfo->pt;
GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
/* get clockrate */
if ((clock_rate = get_clock_rate (src, pt)) == -1)
goto no_clock_rate;
rtptime = pinfo->rtptime;
/* 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 (running_time, clock_rate, GST_SECOND);
/* transit time is difference with RTP timestamp */
transit = rtparrival - rtptime;
/* get ABS diff with previous transit time */
if (src->stats.transit != -1) {
if (transit > src->stats.transit)
diff = transit - src->stats.transit;
else
diff = src->stats.transit - transit;
} else
diff = 0;
src->stats.transit = transit;
/* update jitter, the value we store is scaled up so we can keep precision. */
src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
src->stats.prev_rtptime = src->stats.last_rtptime;
src->stats.last_rtptime = rtparrival;
GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
return;
/* ERRORS */
no_time:
{
GST_WARNING ("cannot get current running_time");
return;
}
no_clock_rate:
{
GST_WARNING ("cannot get clock-rate for pt %d", pt);
return;
}
}
static void
init_seq (RTPSource * src, guint16 seq)
{
src->stats.base_seq = seq;
src->stats.max_seq = seq;
src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
src->stats.cycles = 0;
src->stats.packets_received = 0;
src->stats.octets_received = 0;
src->stats.bytes_received = 0;
src->stats.prev_received = 0;
src->stats.prev_expected = 0;
src->stats.recv_pli_count = 0;
src->stats.recv_fir_count = 0;
GST_DEBUG ("base_seq %d", seq);
}
#define BITRATE_INTERVAL (2 * GST_SECOND)
static void
do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
guint64 * bytes_handled)
{
guint64 elapsed;
if (src->prev_rtime) {
elapsed = running_time - src->prev_rtime;
if (elapsed > BITRATE_INTERVAL) {
guint64 rate;
rate = gst_util_uint64_scale (*bytes_handled, 8 * GST_SECOND, elapsed);
GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
if (src->bitrate == 0)
src->bitrate = rate;
else
src->bitrate = ((src->bitrate * 3) + rate) / 4;
src->prev_rtime = running_time;
*bytes_handled = 0;
}
} else {
GST_LOG ("Reset bitrate measurement");
src->prev_rtime = running_time;
src->bitrate = 0;
}
}
static gboolean
update_receiver_stats (RTPSource * src, RTPPacketInfo * pinfo,
gboolean is_receive)
{
guint16 seqnr, expected;
RTPSourceStats *stats;
gint16 delta;
gint32 packet_rate, max_dropout, max_misorder;
stats = &src->stats;
seqnr = pinfo->seqnum;
packet_rate =
gst_rtp_packet_rate_ctx_update (&src->packet_rate_ctx, pinfo->seqnum,
pinfo->rtptime);
max_dropout =
gst_rtp_packet_rate_ctx_get_max_dropout (&src->packet_rate_ctx,
src->max_dropout_time);
max_misorder =
gst_rtp_packet_rate_ctx_get_max_misorder (&src->packet_rate_ctx,
src->max_misorder_time);
GST_TRACE ("SSRC %08x, packet_rate: %d, max_dropout: %d, max_misorder: %d",
src->ssrc, packet_rate, max_dropout, max_misorder);
if (stats->cycles == -1) {
GST_DEBUG ("received first packet");
/* first time we heard of this source */
init_seq (src, seqnr);
src->stats.max_seq = seqnr - 1;
src->curr_probation = src->probation;
}
if (is_receive) {
expected = src->stats.max_seq + 1;
delta = gst_rtp_buffer_compare_seqnum (expected, seqnr);
/* if we are still on probation, check seqnum */
if (src->curr_probation) {
/* when in probation, we require consecutive seqnums */
if (delta == 0) {
/* expected packet */
GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
src->curr_probation--;
if (seqnr < stats->max_seq) {
/* sequence number wrapped - count another 64K cycle. */
stats->cycles += RTP_SEQ_MOD;
}
src->stats.max_seq = seqnr;
if (src->curr_probation == 0) {
GST_DEBUG ("probation done!");
init_seq (src, seqnr);
} else {
GstBuffer *q;
GST_DEBUG ("probation %d: queue packet", src->curr_probation);
/* when still in probation, keep packets in a list. */
g_queue_push_tail (src->packets, pinfo->data);
pinfo->data = NULL;
/* remove packets from queue if there are too many */
while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
q = g_queue_pop_head (src->packets);
gst_buffer_unref (q);
}
goto done;
}
} else {
/* unexpected seqnum in probation
*
* There is no need to clean the queue at this point because the
* invalid packets in the queue are not going to be pushed as we are
* still in probation, and some cleanup will be performed at future
* probation attempts anyway if there are too many old packets in the
* queue.
*/
goto probation_seqnum;
}
} else if (delta >= 0 && delta < max_dropout) {
/* Clear bad packets */
stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
g_queue_clear (src->packets);
/* in order, with permissible gap */
if (seqnr < stats->max_seq) {
/* sequence number wrapped - count another 64K cycle. */
stats->cycles += RTP_SEQ_MOD;
}
stats->max_seq = seqnr;
} else if (delta < -max_misorder || delta >= max_dropout) {
/* the sequence number made a very large jump */
if (seqnr == stats->bad_seq && src->packets->head) {
/* two sequential packets -- assume that the other side
* restarted without telling us so just re-sync
* (i.e., pretend this was the first packet). */
init_seq (src, seqnr);
} else {
/* unacceptable jump */
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
g_queue_clear (src->packets);
g_queue_push_tail (src->packets, pinfo->data);
pinfo->data = NULL;
goto bad_sequence;
}
} else { /* delta < 0 && delta >= -max_misorder */
/* Clear bad packets */
stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
g_queue_clear (src->packets);
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
GST_INFO ("duplicate or reordered packet (seqnr %u, expected %u)",
seqnr, expected);
}
}
src->stats.octets_received += pinfo->payload_len;
src->stats.bytes_received += pinfo->bytes;
src->stats.packets_received += pinfo->packets;
/* for the bitrate estimation */
src->bytes_received += pinfo->payload_len;
GST_LOG ("seq %u, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
seqnr, src->stats.packets_received, src->stats.octets_received);
return TRUE;
/* ERRORS */
done:
{
return FALSE;
}
bad_sequence:
{
GST_WARNING
("unacceptable seqnum received (seqnr %u, delta %d, packet_rate: %d, max_dropout: %d, max_misorder: %d)",
seqnr, delta, packet_rate, max_dropout, max_misorder);
return FALSE;
}
probation_seqnum:
{
GST_WARNING ("probation: seqnr %d != expected %d", seqnr, expected);
src->curr_probation = src->probation;
src->stats.max_seq = seqnr;
return FALSE;
}
}
/**
* rtp_source_process_rtp:
* @src: an #RTPSource
* @pinfo: an #RTPPacketInfo
*
* Let @src handle the incoming RTP packet described in @pinfo.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo)
{
GstFlowReturn result;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (pinfo != NULL, GST_FLOW_ERROR);
if (!update_receiver_stats (src, pinfo, TRUE))
return GST_FLOW_OK;
/* the source that sent the packet must be a sender */
src->is_sender = TRUE;
src->validated = TRUE;
do_bitrate_estimation (src, pinfo->running_time, &src->bytes_received);
/* calculate jitter for the stats */
calculate_jitter (src, pinfo);
/* we're ready to push the RTP packet now */
result = push_packet (src, pinfo->data);
pinfo->data = NULL;
return result;
}
/**
* rtp_source_mark_bye:
* @src: an #RTPSource
* @reason: the reason for leaving
*
* Mark @src in the BYE state. This can happen when the source wants to
* leave the sesssion or when a BYE packets has been received.
*
* This will make the source inactive.
*/
void
rtp_source_mark_bye (RTPSource * src, const gchar * reason)
{
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
GST_STR_NULL (reason));
/* copy the reason and mark as bye */
g_free (src->bye_reason);
src->bye_reason = g_strdup (reason);
src->marked_bye = TRUE;
}
/**
* rtp_source_send_rtp:
* @src: an #RTPSource
* @pinfo: an #RTPPacketInfo
*
* Send data (an RTP buffer or buffer list from @pinfo) originating from @src.
* This will make @src a sender. This function takes ownership of the data and
* modifies the SSRC in the RTP packet to that of @src when needed.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
{
GstFlowReturn result;
GstClockTime running_time;
guint32 rtptime;
guint64 ext_rtptime;
guint64 rt_diff, rtp_diff;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
/* we are a sender now */
src->is_sender = TRUE;
/* we are also a receiver of our packets */
if (!update_receiver_stats (src, pinfo, FALSE))
return GST_FLOW_OK;
if (src->pt_set && src->pt != pinfo->pt) {
GST_WARNING ("Changing pt from %u to %u for SSRC %u", src->pt, pinfo->pt,
src->ssrc);
}
src->pt = pinfo->pt;
src->pt_set = TRUE;
/* update stats for the SR */
src->stats.packets_sent += pinfo->packets;
src->stats.octets_sent += pinfo->payload_len;
src->bytes_sent += pinfo->payload_len;
running_time = pinfo->running_time;
do_bitrate_estimation (src, running_time, &src->bytes_sent);
rtptime = pinfo->rtptime;
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", running_time %"
GST_TIME_FORMAT, src->ssrc, ext_rtptime, GST_TIME_ARGS (running_time));
if (ext_rtptime > src->last_rtptime) {
rtp_diff = ext_rtptime - src->last_rtptime;
rt_diff = running_time - src->last_rtime;
/* calc the diff so we can detect drift at the sender. This can also be used
* to guestimate the clock rate if the NTP time is locked to the RTP
* timestamps (as is the case when the capture device is providing the clock). */
GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff running_time %"
GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (rt_diff));
}
/* we keep track of the last received RTP timestamp and the corresponding
* buffer running_time so that we can use this info when constructing SR reports */
src->last_rtime = running_time;
src->last_rtptime = ext_rtptime;
/* push packet */
if (!src->callbacks.push_rtp)
goto no_callback;
GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT,
pinfo->is_list ? "list" : "packet", src->stats.packets_sent);
result = src->callbacks.push_rtp (src, pinfo->data, src->user_data);
pinfo->data = NULL;
return result;
/* ERRORS */
no_callback:
{
GST_WARNING ("no callback installed, dropping packet");
return GST_FLOW_OK;
}
}
/**
* rtp_source_process_sr:
* @src: an #RTPSource
* @time: time of packet arrival
* @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
* @rtptime: the RTP time (in clock rate units)
* @packet_count: the packet count
* @octet_count: the octet count
*
* Update the sender report in @src.
*/
void
rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
guint32 rtptime, guint32 packet_count, guint32 octet_count)
{
RTPSenderReport *curr;
gint curridx;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
(guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
packet_count, octet_count);
curridx = src->stats.curr_sr ^ 1;
curr = &src->stats.sr[curridx];
/* this is a sender now */
src->is_sender = TRUE;
/* update current */
curr->is_valid = TRUE;
curr->ntptime = ntptime;
curr->rtptime = rtptime;
curr->packet_count = packet_count;
curr->octet_count = octet_count;
curr->time = time;
/* make current */
src->stats.curr_sr = curridx;
src->stats.prev_rtcptime = src->stats.last_rtcptime;
src->stats.last_rtcptime = time;
}
/**
* rtp_source_process_rb:
* @src: an #RTPSource
* @ntpnstime: the current time in nanoseconds since 1970
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumulative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter (in clock rate units)
* @lsr: the time of the last SR packet on this source
* (in NTP Short Format, 16.16 fixed point)
* @dlsr: the delay since the last SR packet
* (in NTP Short Format, 16.16 fixed point)
*
* Update the report block in @src.
*/
void
rtp_source_process_rb (RTPSource * src, guint64 ntpnstime,
guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
guint32 jitter, guint32 lsr, guint32 dlsr)
{
RTPReceiverReport *curr;
gint curridx;
guint32 ntp, A;
guint64 f_ntp;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
curridx = src->stats.curr_rr ^ 1;
curr = &src->stats.rr[curridx];
/* update current */
curr->is_valid = TRUE;
curr->fractionlost = fractionlost;
curr->packetslost = packetslost;
curr->exthighestseq = exthighestseq;
curr->jitter = jitter;
curr->lsr = lsr;
curr->dlsr = dlsr;
/* convert the NTP time in nanoseconds to 32.32 fixed point */
f_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
/* calculate round trip, round the time up */
ntp = ((f_ntp + 0xffff) >> 16) & 0xffffffff;
A = dlsr + lsr;
if (A > 0 && ntp > A)
A = ntp - A;
else
A = 0;
curr->round_trip = A;
GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
A >> 16, A & 0xffff);
/* make current */
src->stats.curr_rr = curridx;
}
/**
* rtp_source_get_new_sr:
* @src: an #RTPSource
* @ntpnstime: the current time in nanoseconds since 1970
* @running_time: the current running_time of the pipeline
* @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
* @rtptime: the RTP time corresponding to @ntptime (in clock rate units)
* @packet_count: the packet count
* @octet_count: the octet count
*
* Get new values to put into a new SR report from this source.
*
* @running_time and @ntpnstime are captured at the same time and represent the
* running time of the pipeline clock and the absolute current system time in
* nanoseconds respectively. Together with the last running_time and RTP timestamp
* we have observed in the source, we can generate @ntptime and @rtptime for an SR
* packet. @ntptime is basically the fixed point representation of @ntpnstime
* and @rtptime the associated RTP timestamp.
*
* Returns: %TRUE on success.
*/
gboolean
rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
GstClockTime running_time, guint64 * ntptime, guint32 * rtptime,
guint32 * packet_count, guint32 * octet_count)
{
guint64 t_rtp;
guint64 t_current_ntp;
GstClockTimeDiff diff;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
/* We last saw a buffer with last_rtptime at last_rtime. Given a running_time
* and an NTP time, we can scale the RTP timestamps so that they match the
* given NTP time. for scaling, we assume that the slope of the rtptime vs
* running_time vs ntptime curve is close to 1, which is certainly
* sufficient for the frequency at which we report SR and the rate we send
* out RTP packets. */
t_rtp = src->last_rtptime;
GST_DEBUG ("last_rtime %" GST_TIME_FORMAT ", last_rtptime %"
G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_rtime), t_rtp);
if (src->clock_rate == -1 && src->pt_set) {
GST_INFO ("no clock-rate, getting for pt %u and SSRC %u", src->pt,
src->ssrc);
get_clock_rate (src, src->pt);
}
if (src->clock_rate != -1) {
/* get the diff between the clock running_time and the buffer running_time.
* This is the elapsed time, as measured against the pipeline clock, between
* when the rtp timestamp was observed and the current running_time.
*
* We need to apply this diff to the RTP timestamp to get the RTP timestamp
* for the given ntpnstime. */
diff = GST_CLOCK_DIFF (src->last_rtime, running_time);
GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_STIME_FORMAT,
GST_TIME_ARGS (running_time), GST_STIME_ARGS (diff));
/* now translate the diff to RTP time, handle positive and negative cases.
* If there is no diff, we already set rtptime correctly above. */
if (diff > 0) {
t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
} else {
diff = -diff;
t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
}
} else {
GST_WARNING ("no clock-rate, cannot interpolate rtp time for SSRC %u",
src->ssrc);
}
/* convert the NTP time in nanoseconds to 32.32 fixed point */
t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
(guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
(guint32) t_rtp);
if (ntptime)
*ntptime = t_current_ntp;
if (rtptime)
*rtptime = t_rtp;
if (packet_count)
*packet_count = src->stats.packets_sent;
if (octet_count)
*octet_count = src->stats.octets_sent;
return TRUE;
}
/**
* rtp_source_get_new_rb:
* @src: an #RTPSource
* @time: the current time of the system clock
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumulative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter (in clock rate units)
* @lsr: the time of the last SR packet on this source
* (in NTP Short Format, 16.16 fixed point)
* @dlsr: the delay since the last SR packet
* (in NTP Short Format, 16.16 fixed point)
*
* Get new values to put into a new report block from this source.
*
* Returns: %TRUE on success.
*/
gboolean
rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
guint32 * jitter, guint32 * lsr, guint32 * dlsr)
{
RTPSourceStats *stats;
guint64 extended_max, expected;
guint64 expected_interval, received_interval, ntptime;
gint64 lost, lost_interval;
guint32 fraction, LSR, DLSR;
GstClockTime sr_time;
stats = &src->stats;
extended_max = stats->cycles + stats->max_seq;
expected = extended_max - stats->base_seq + 1;
GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
extended_max, expected, stats->packets_received, stats->base_seq);
lost = expected - stats->packets_received;
lost = CLAMP (lost, -0x800000, 0x7fffff);
expected_interval = expected - stats->prev_expected;
stats->prev_expected = expected;
received_interval = stats->packets_received - stats->prev_received;
stats->prev_received = stats->packets_received;
lost_interval = expected_interval - received_interval;
if (expected_interval == 0 || lost_interval <= 0)
fraction = 0;
else
fraction = (lost_interval << 8) / expected_interval;
GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
/* we scaled the jitter up for additional precision */
GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
extended_max, stats->jitter >> 4);
if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
GstClockTime diff;
/* LSR is middle 32 bits of the last ntptime */
LSR = (ntptime >> 16) & 0xffffffff;
diff = time - sr_time;
GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
/* DLSR, delay since last SR is expressed in 1/65536 second units */
DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
} else {
/* No valid SR received, LSR/DLSR are set to 0 then */
GST_DEBUG ("no valid SR received");
LSR = 0;
DLSR = 0;
}
GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
DLSR >> 16, DLSR & 0xffff);
if (fractionlost)
*fractionlost = fraction;
if (packetslost)
*packetslost = lost;
if (exthighestseq)
*exthighestseq = extended_max;
if (jitter)
*jitter = stats->jitter >> 4;
if (lsr)
*lsr = LSR;
if (dlsr)
*dlsr = DLSR;
return TRUE;
}
/**
* rtp_source_get_last_sr:
* @src: an #RTPSource
* @time: time of packet arrival
* @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
* @rtptime: the RTP time (in clock rate units)
* @packet_count: the packet count
* @octet_count: the octet count
*
* Get the values of the last sender report as set with rtp_source_process_sr().
*
* Returns: %TRUE if there was a valid SR report.
*/
gboolean
rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
{
RTPSenderReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.sr[src->stats.curr_sr];
if (!curr->is_valid)
return FALSE;
if (ntptime)
*ntptime = curr->ntptime;
if (rtptime)
*rtptime = curr->rtptime;
if (packet_count)
*packet_count = curr->packet_count;
if (octet_count)
*octet_count = curr->octet_count;
if (time)
*time = curr->time;
return TRUE;
}
/**
* rtp_source_get_last_rb:
* @src: an #RTPSource
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumulative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter (in clock rate units)
* @lsr: the time of the last SR packet on this source
* (in NTP Short Format, 16.16 fixed point)
* @dlsr: the delay since the last SR packet
* (in NTP Short Format, 16.16 fixed point)
* @round_trip: the round-trip time
* (in NTP Short Format, 16.16 fixed point)
*
* Get the values of the last RB report set with rtp_source_process_rb().
*
* Returns: %TRUE if there was a valid SB report.
*/
gboolean
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
{
RTPReceiverReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.rr[src->stats.curr_rr];
if (!curr->is_valid)
return FALSE;
if (fractionlost)
*fractionlost = curr->fractionlost;
if (packetslost)
*packetslost = curr->packetslost;
if (exthighestseq)
*exthighestseq = curr->exthighestseq;
if (jitter)
*jitter = curr->jitter;
if (lsr)
*lsr = curr->lsr;
if (dlsr)
*dlsr = curr->dlsr;
if (round_trip)
*round_trip = curr->round_trip;
return TRUE;
}
gboolean
find_conflicting_address (GList * conflicting_addresses,
GSocketAddress * address, GstClockTime time)
{
GList *item;
for (item = conflicting_addresses; item; item = g_list_next (item)) {
RTPConflictingAddress *known_conflict = item->data;
if (__g_socket_address_equal (address, known_conflict->address)) {
known_conflict->time = time;
return TRUE;
}
}
return FALSE;
}
GList *
add_conflicting_address (GList * conflicting_addresses,
GSocketAddress * address, GstClockTime time)
{
RTPConflictingAddress *new_conflict;
new_conflict = g_slice_new (RTPConflictingAddress);
new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address));
new_conflict->time = time;
return g_list_prepend (conflicting_addresses, new_conflict);
}
GList *
timeout_conflicting_addresses (GList * conflicting_addresses,
GstClockTime current_time)
{
GList *item;
/* "a relatively long time" -- RFC 3550 section 8.2 */
const GstClockTime collision_timeout =
RTP_STATS_MIN_INTERVAL * GST_SECOND * 10;
item = g_list_first (conflicting_addresses);
while (item) {
RTPConflictingAddress *known_conflict = item->data;
GList *next_item = g_list_next (item);
if (known_conflict->time < current_time - collision_timeout) {
gchar *buf;
conflicting_addresses = g_list_delete_link (conflicting_addresses, item);
buf = __g_socket_address_to_string (known_conflict->address);
GST_DEBUG ("collision %p timed out: %s", known_conflict, buf);
g_free (buf);
rtp_conflicting_address_free (known_conflict);
}
item = next_item;
}
return conflicting_addresses;
}
/**
* rtp_source_find_conflicting_address:
* @src: The source the packet came in
* @address: address to check for
* @time: The time when the packet that is possibly in conflict arrived
*
* Checks if an address which has a conflict is already known. If it is
* a known conflict, remember the time
*
* Returns: TRUE if it was a known conflict, FALSE otherwise
*/
gboolean
rtp_source_find_conflicting_address (RTPSource * src, GSocketAddress * address,
GstClockTime time)
{
return find_conflicting_address (src->conflicting_addresses, address, time);
}
/**
* rtp_source_add_conflicting_address:
* @src: The source the packet came in
* @address: address to remember
* @time: The time when the packet that is in conflict arrived
*
* Adds a new conflict address
*/
void
rtp_source_add_conflicting_address (RTPSource * src,
GSocketAddress * address, GstClockTime time)
{
src->conflicting_addresses =
add_conflicting_address (src->conflicting_addresses, address, time);
}
/**
* rtp_source_timeout:
* @src: The #RTPSource
* @current_time: The current time
* @feedback_retention_window: The running time before which retained feedback
* packets have to be discarded
*
* This is processed on each RTCP interval. It times out old collisions.
* It also times out old retained feedback packets
*/
void
rtp_source_timeout (RTPSource * src, GstClockTime current_time,
GstClockTime running_time, GstClockTime feedback_retention_window)
{
GstRTCPPacket *pkt;
GstClockTime max_pts_window;
guint pruned = 0;
src->conflicting_addresses =
timeout_conflicting_addresses (src->conflicting_addresses, current_time);
if (feedback_retention_window == GST_CLOCK_TIME_NONE ||
running_time < feedback_retention_window) {
return;
}
max_pts_window = running_time - feedback_retention_window;
/* Time out AVPF packets that are older than the desired length */
while ((pkt = g_queue_peek_head (src->retained_feedback)) &&
GST_BUFFER_PTS (pkt) < max_pts_window) {
gst_buffer_unref (g_queue_pop_head (src->retained_feedback));
pruned++;
}
GST_LOG_OBJECT (src,
"%u RTCP packets pruned with PTS less than %" GST_TIME_FORMAT
", queue len: %u", pruned, GST_TIME_ARGS (max_pts_window),
g_queue_get_length (src->retained_feedback));
}
static gint
compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data)
{
const GstBuffer *bufa = a;
const GstBuffer *bufb = b;
g_return_val_if_fail (GST_BUFFER_PTS (bufa) != GST_CLOCK_TIME_NONE, -1);
g_return_val_if_fail (GST_BUFFER_PTS (bufb) != GST_CLOCK_TIME_NONE, 1);
if (GST_BUFFER_PTS (bufa) < GST_BUFFER_PTS (bufb)) {
return -1;
} else if (GST_BUFFER_PTS (bufa) > GST_BUFFER_PTS (bufb)) {
return 1;
}
return 0;
}
void
rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet,
GstClockTime running_time)
{
GstBuffer *buffer;
g_return_if_fail (running_time != GST_CLOCK_TIME_NONE);
buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY,
packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4);
GST_BUFFER_PTS (buffer) = running_time;
g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL);
GST_LOG_OBJECT (src, "RTCP packet retained with PTS: %" GST_TIME_FORMAT,
GST_TIME_ARGS (running_time));
}
gboolean
rtp_source_has_retained (RTPSource * src, GCompareFunc func, gconstpointer data)
{
if (g_queue_find_custom (src->retained_feedback, data, func))
return TRUE;
else
return FALSE;
}
/**
* rtp_source_register_nack:
* @src: The #RTPSource
* @seqnum: a seqnum
* @deadline: the deadline before which RTX is still possible
*
* Register that @seqnum has not been received from @src.
*/
void
rtp_source_register_nack (RTPSource * src, guint16 seqnum,
GstClockTime deadline)
{
gint i;
guint len;
gint diff = -1;
guint16 tseq;
len = src->nacks->len;
for (i = len - 1; i >= 0; i--) {
tseq = g_array_index (src->nacks, guint16, i);
diff = gst_rtp_buffer_compare_seqnum (tseq, seqnum);
GST_TRACE ("[%u] %u %u diff %i len %u", i, tseq, seqnum, diff, len);
if (diff >= 0)
break;
}
if (diff == 0) {
GST_DEBUG ("update NACK #%u deadline to %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (deadline));
g_array_index (src->nack_deadlines, GstClockTime, i) = deadline;
} else if (i == len - 1) {
GST_DEBUG ("append NACK #%u with deadline %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (deadline));
g_array_append_val (src->nacks, seqnum);
g_array_append_val (src->nack_deadlines, deadline);
} else {
GST_DEBUG ("insert NACK #%u with deadline %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (deadline));
g_array_insert_val (src->nacks, i + 1, seqnum);
g_array_insert_val (src->nack_deadlines, i + 1, deadline);
}
src->send_nack = TRUE;
}
/**
* rtp_source_get_nacks:
* @src: The #RTPSource
* @n_nacks: result number of nacks
*
* Get the registered NACKS since the last rtp_source_clear_nacks().
*
* Returns: an array of @n_nacks seqnum values.
*/
guint16 *
rtp_source_get_nacks (RTPSource * src, guint * n_nacks)
{
if (n_nacks)
*n_nacks = src->nacks->len;
return (guint16 *) src->nacks->data;
}
/**
* rtp_source_get_nacks:
* @src: The #RTPSource
* @n_nacks: result number of nacks
*
* Get the registered NACKS deadlines.
*
* Returns: an array of @n_nacks deadline values.
*/
GstClockTime *
rtp_source_get_nack_deadlines (RTPSource * src, guint * n_nacks)
{
if (n_nacks)
*n_nacks = src->nack_deadlines->len;
return (GstClockTime *) src->nack_deadlines->data;
}
/**
* rtp_source_clear_nacks:
* @src: The #RTPSource
* @n_nacks: number of nacks
*
* Remove @n_nacks oldest NACKS form array.
*/
void
rtp_source_clear_nacks (RTPSource * src, guint n_nacks)
{
g_return_if_fail (n_nacks <= src->nacks->len);
if (src->nacks->len == n_nacks) {
g_array_set_size (src->nacks, 0);
g_array_set_size (src->nack_deadlines, 0);
src->send_nack = FALSE;
} else {
g_array_remove_range (src->nacks, 0, n_nacks);
g_array_remove_range (src->nack_deadlines, 0, n_nacks);
}
}
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