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gstreamer/subprojects/gst-plugins-good/gst/rtpmanager/rtpsource.c

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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 payload 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 payload bytes received
* * "packets-received" G_TYPE_UINT64 total number of packets received
* * "bytes-received" G_TYPE_UINT64 total number of bytes received including lower level headers overhead
*
* 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));
/**
* RTPSource: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_free (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;
guint32 ssrc = 0;
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,
"bytes-received", G_TYPE_UINT64, src->stats.bytes_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,
"recv-packet-rate", G_TYPE_UINT,
gst_rtp_packet_rate_ctx_get (&src->packet_rate_ctx), 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, &ssrc, &fractionlost,
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
gst_structure_set (s,
"have-rb", G_TYPE_BOOLEAN, have_rb,
"rb-ssrc", G_TYPE_UINT, ssrc,
"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.caps = cb->caps;
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_send_caps (RTPSource * src, GstCaps * caps)
{
GstStructure *s;
guint val;
gint ival;
guint ssrc, rtx_ssrc = -1;
gboolean rtx;
/* nothing changed, return */
if (caps == NULL || src->caps == caps)
return;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
return;
gst_structure_get_uint (s, "rtx-ssrc", &rtx_ssrc);
if (src->ssrc != ssrc && src->ssrc != rtx_ssrc) {
GST_WARNING ("got ssrc %u/%u that doesn't match with this source's ssrc %u",
ssrc, rtx_ssrc, src->ssrc);
return;
}
rtx = (rtx_ssrc == 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);
if (rtx) {
src->media_ssrc = ssrc;
} else {
src->media_ssrc = -1;
}
}
/**
* 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 void
fetch_caps_for_payload (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 || !src->caps) {
GstCaps *caps = NULL;
if (src->callbacks.caps) {
caps = src->callbacks.caps (src, payload, src->user_data);
}
GST_DEBUG ("got caps %" GST_PTR_FORMAT, caps);
if (caps) {
const GstStructure *s;
gint clock_rate = -1;
const gchar *encoding_name;
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "clock-rate", &clock_rate)) {
src->clock_rate = clock_rate;
gst_rtp_packet_rate_ctx_reset (&src->packet_rate_ctx, clock_rate);
} else {
GST_DEBUG ("No clock-rate in caps!");
}
encoding_name = gst_structure_get_string (s, "encoding-name");
/* Disable probation for RTX sources as packets will arrive very
* irregularly and waiting for a second packet usually exceeds the
* deadline of the retransmission */
if (g_strcmp0 (encoding_name, "rtx") == 0) {
src->probation = src->curr_probation = 0;
}
}
gst_caps_replace (&src->caps, caps);
gst_clear_caps (&caps);
}
}
/* 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;
/* get arrival time */
if ((running_time = pinfo->running_time) == GST_CLOCK_TIME_NONE)
goto no_time;
GST_LOG ("SSRC %08x got payload %d", src->ssrc, pinfo->pt);
/* check if clock-rate is valid */
if (src->clock_rate == -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, src->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, src->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", pinfo->pt);
return;
}
}
static void
update_queued_stats (GstBuffer * buffer, RTPSource * src)
{
GstRTPBuffer rtp = { NULL };
guint payload_len;
guint64 bytes;
/* no need to check the return value, a queued packet is a valid RTP one */
gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
bytes = gst_buffer_get_size (buffer) + UDP_IP_HEADER_OVERHEAD;
src->stats.octets_received += payload_len;
src->stats.bytes_received += bytes;
src->stats.packets_received++;
/* for the bitrate estimation consider all lower level headers */
src->bytes_received += bytes;
gst_rtp_buffer_unmap (&rtp);
}
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;
/* if there are queued packets, consider them too in the stats */
g_queue_foreach (src->packets, (GFunc) update_queued_stats, src);
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);
}
} else {
/* Sender stats - update the outbound sequence number */
expected = src->stats.max_seq + 1;
delta = gst_rtp_buffer_compare_seqnum (expected, seqnr);
/* No probation for local senders, just check for lost / dropouts */
if (delta >= 0 && delta < max_dropout) {
stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
/* 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) {
/* unacceptable jump */
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
}
} else { /* delta < 0 && delta >= -max_misorder */
stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
}
}
src->stats.octets_received += pinfo->payload_len;
src->stats.bytes_received += pinfo->bytes;
src->stats.packets_received += pinfo->packets;
/* for the bitrate estimation consider all lower level headers */
src->bytes_received += pinfo->bytes;
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 "
"(SSRC %u curr_probation %i probation %i)", seqnr, expected, src->ssrc,
src->curr_probation, src->probation);
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);
fetch_caps_for_payload (src, pinfo->pt);
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 session 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->bytes;
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
* @ssrc: SSRC of the local source for this this RB was sent
* @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, guint32 ssrc, 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->ssrc = ssrc;
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);
fetch_caps_for_payload (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
* @ssrc: SSRC of the local source for this this RB was sent
* @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, guint32 * ssrc,
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 (ssrc)
*ssrc = curr->ssrc;
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_new (RTPConflictingAddress, 1);
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 + collision_timeout < current_time) {
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_nack_deadlines:
* @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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