/* 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
*
* 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., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
#include <string.h>
#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_IS_SENDER,
PROP_SDES,
PROP_STATS,
- PROP_LAST
+ PROP_PROBATION,
+ PROP_MAX_DROPOUT_TIME,
+ PROP_MAX_MISORDER_TIME,
+ PROP_DISABLE_RTCP
};
/* GObject vmethods */
* 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
+ * '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
* '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
+ * 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,
/**
* RTPSource::stats
*
- * The statistics of the source. This property returns a GstStructure with
- * name application/x-rtp-source-stats with the following fields:
+ * This property returns a GstStructure named application/x-rtp-source-stats with
+ * fields useful for statistics and diagnostics.
*
- * "ssrc" G_TYPE_UINT The SSRC of this source
- * "internal" G_TYPE_BOOLEAN If this source is the source of the session
- * "validated" G_TYPE_BOOLEAN If the source is validated
- * "received-bye" G_TYPE_BOOLEAN If we received a BYE from this source
- * "is-csrc" G_TYPE_BOOLEAN If this source was found as CSRC
- * "is-sender" G_TYPE_BOOLEAN If this source is a sender
+ * 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 two fields are only present when known.
+ * 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:
+ * 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
* Following fields are updated when "is-sender" is TRUE.
*
* "bitrate" G_TYPE_UINT64 bitrate in bits per second
- * "jitter" G_TYPE_UINT estimated jitter
+ * "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 ntptime of SR
- * "sr-rtptime" G_TYPE_UINT rtptime of 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
*
* 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 fraction
+ * "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
- * "sent-rb-lsr" G_TYPE_UINT last SR time
- * "sent-rb-dlsr" G_TYPE_UINT delay since last SR
+ * "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 fraction
+ * "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
- * "rb-lsr" G_TYPE_UINT last SR time
- * "rb-dlsr" G_TYPE_UINT delay since last SR
+ * "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. This is calculated from the last RB
- * values and the recption time of the last RB packet. Only present for
+ * 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 in nanoseconds
+ * "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");
}
void
rtp_source_reset (RTPSource * src)
{
- src->received_bye = FALSE;
+ 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 initialy on probation until we receive enough valid RTP
+ /* 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 = RTP_DEFAULT_PROBATION;
+ 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_base = -1;
- src->last_rtptime = -1;
+ 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;
}
-static void
+void
rtp_conflicting_address_free (RTPConflictingAddress * addr)
{
g_object_unref (addr->address);
- g_free (addr);
+ g_slice_free (RTPConflictingAddress, addr);
}
static void
rtp_source_finalize (GObject * object)
{
RTPSource *src;
- GstBuffer *buffer;
src = RTP_SOURCE_CAST (object);
- while ((buffer = g_queue_pop_head (src->packets)))
- gst_buffer_unref (buffer);
+ g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
g_queue_free (src->packets);
gst_structure_free (src->sdes);
gst_caps_replace (&src->caps, NULL);
- g_list_foreach (src->conflicting_addresses,
- (GFunc) rtp_conflicting_address_free, NULL);
- g_list_free (src->conflicting_addresses);
-
- while ((buffer = g_queue_pop_head (src->retained_feedback)))
- gst_buffer_unref (buffer);
+ 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);
}
"ssrc", G_TYPE_UINT, (guint) src->ssrc,
"internal", G_TYPE_BOOLEAN, internal,
"validated", G_TYPE_BOOLEAN, src->validated,
- "received-bye", G_TYPE_BOOLEAN, src->received_bye,
+ "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_base,
+ "seqnum-base", G_TYPE_INT, src->seqnum_offset,
"clock-rate", G_TYPE_INT, src->clock_rate, NULL);
/* add address and port */
"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), NULL);
+ (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,
}
/**
- * rtp_source_set_sdes:
+ * rtp_source_set_sdes_struct:
* @src: an #RTPSource
* @sdes: the SDES structure
*
} else {
gst_structure_free (sdes);
}
-
return changed;
}
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;
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_received_bye:
+ * rtp_source_is_marked_bye:
* @src: an #RTPSource
*
- * Check if @src has receoved a BYE packet.
+ * Check if @src is marked as leaving the session with a BYE packet.
*
- * Returns: %TRUE if @src has received a BYE packet.
+ * Returns: %TRUE if @src has been marked BYE.
*/
gboolean
-rtp_source_received_bye (RTPSource * src)
+rtp_source_is_marked_bye (RTPSource * src)
{
gboolean result;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
- result = src->received_bye;
+ 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 receoved a BYE message first
- * with rtp_source_received_bye().
+ * 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 did
- * not receive a BYE message yet. g_fee() after usage.
+ * 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)
GstStructure *s;
guint val;
gint ival;
+ gboolean rtx;
/* nothing changed, return */
if (caps == NULL || src->caps == caps)
s = gst_caps_get_structure (caps, 0);
- if (gst_structure_get_int (s, "payload", &ival))
+ 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 payload %d", src->payload);
+
+ GST_DEBUG ("got %spayload %d", rtx ? "rtx " : "", src->payload);
if (gst_structure_get_int (s, "clock-rate", &ival))
src->clock_rate = ival;
GST_DEBUG ("got clock-rate %d", src->clock_rate);
- if (gst_structure_get_uint (s, "seqnum-base", &val))
- src->seqnum_base = val;
+ if (gst_structure_get_uint (s, rtx ? "rtx-seqnum-offset" : "seqnum-offset",
+ &val))
+ src->seqnum_offset = val;
else
- src->seqnum_base = -1;
+ src->seqnum_offset = -1;
- GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
+ GST_DEBUG ("got %sseqnum-offset %" G_GINT32_FORMAT, rtx ? "rtx " : "",
+ src->seqnum_offset);
gst_caps_replace (&src->caps, caps);
}
/**
- * rtp_source_set_sdes_string:
- * @src: an #RTPSource
- * @type: the type of the SDES item
- * @data: the SDES data
- *
- * Store an SDES item of @type in @src.
- *
- * Returns: %FALSE if the SDES item was unchanged or @type is unknown.
- */
-gboolean
-rtp_source_set_sdes_string (RTPSource * src, GstRTCPSDESType type,
- const gchar * data)
-{
- const gchar *old;
- const gchar *field;
-
- field = gst_rtcp_sdes_type_to_name (type);
-
- if (gst_structure_has_field (src->sdes, field))
- old = gst_structure_get_string (src->sdes, field);
- else
- old = NULL;
-
- if (old == NULL && data == NULL)
- return FALSE;
-
- if (old != NULL && data != NULL && strcmp (old, data) == 0)
- return FALSE;
-
- if (data == NULL)
- gst_structure_remove_field (src->sdes, field);
- else
- gst_structure_set (src->sdes, field, G_TYPE_STRING, data, NULL);
-
- return TRUE;
-}
-
-/**
- * rtp_source_get_sdes_string:
- * @src: an #RTPSource
- * @type: the type of the SDES item
- *
- * Get the SDES item of @type from @src.
- *
- * Returns: a null-terminated copy of the SDES item or NULL when @type was not
- * valid or the SDES item was unset. g_free() after usage.
- */
-gchar *
-rtp_source_get_sdes_string (RTPSource * src, GstRTCPSDESType type)
-{
- gchar *result;
- const gchar *type_name;
-
- g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
-
- if (type < 0 || type > GST_RTCP_SDES_PRIV - 1)
- return NULL;
-
- type_name = gst_rtcp_sdes_type_to_name (type);
-
- if (!gst_structure_has_field (src->sdes, type_name))
- return NULL;
-
- result = g_strdup (gst_structure_get_string (src->sdes, type_name));
-
- return result;
-}
-
-/**
* rtp_source_set_rtp_from:
* @src: an #RTPSource
* @address: the RTP address to set
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;
}
* 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
* milliseconds. */
static void
-calculate_jitter (RTPSource * src, GstBuffer * buffer,
- RTPArrivalStats * arrival)
+calculate_jitter (RTPSource * src, RTPPacketInfo * pinfo)
{
GstClockTime running_time;
guint32 rtparrival, transit, rtptime;
gint32 diff;
gint clock_rate;
guint8 pt;
- GstRTPBuffer rtp = { NULL };
/* get arrival time */
- if ((running_time = arrival->running_time) == GST_CLOCK_TIME_NONE)
+ if ((running_time = pinfo->running_time) == GST_CLOCK_TIME_NONE)
goto no_time;
- gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
- pt = gst_rtp_buffer_get_payload_type (&rtp);
+ pt = pinfo->pt;
GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
/* get clockrate */
- if ((clock_rate = get_clock_rate (src, pt)) == -1) {
- gst_rtp_buffer_unmap (&rtp);
+ if ((clock_rate = get_clock_rate (src, pt)) == -1)
goto no_clock_rate;
- }
- rtptime = gst_rtp_buffer_get_timestamp (&rtp);
+ 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. */
GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
- gst_rtp_buffer_unmap (&rtp);
return;
/* ERRORS */
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);
}
}
}
-/**
- * rtp_source_process_rtp:
- * @src: an #RTPSource
- * @buffer: an RTP buffer
- *
- * Let @src handle the incomming RTP @buffer.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
- RTPArrivalStats * arrival)
+static gboolean
+update_receiver_stats (RTPSource * src, RTPPacketInfo * pinfo,
+ gboolean is_receive)
{
- GstFlowReturn result = GST_FLOW_OK;
- guint16 seqnr, udelta;
+ guint16 seqnr, expected;
RTPSourceStats *stats;
- guint16 expected;
- GstRTPBuffer rtp = { NULL };
-
- g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
- g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ gint16 delta;
+ gint32 packet_rate, max_dropout, max_misorder;
stats = &src->stats;
- gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
- seqnr = gst_rtp_buffer_get_seq (&rtp);
- gst_rtp_buffer_unmap (&rtp);
+ 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 buffer");
+ GST_DEBUG ("received first packet");
/* first time we heard of this source */
init_seq (src, seqnr);
src->stats.max_seq = seqnr - 1;
- src->probation = RTP_DEFAULT_PROBATION;
+ src->curr_probation = src->probation;
}
- udelta = seqnr - stats->max_seq;
-
- /* if we are still on probation, check seqnum */
- if (src->probation) {
+ if (is_receive) {
expected = src->stats.max_seq + 1;
-
- /* when in probation, we require consecutive seqnums */
- if (seqnr == expected) {
- /* expected packet */
- GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
- src->probation--;
- src->stats.max_seq = seqnr;
- if (src->probation == 0) {
- GST_DEBUG ("probation done!");
+ 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 {
- GstBuffer *q;
-
- GST_DEBUG ("probation %d: queue buffer", src->probation);
- /* when still in probation, keep packets in a list. */
- g_queue_push_tail (src->packets, buffer);
- /* 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;
+ /* 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 {
- /* unexpected seqnum in probation */
- goto probation_seqnum;
- }
- } else if (udelta < RTP_MAX_DROPOUT) {
- /* 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 (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
- /* the sequence number made a very large jump */
- if (seqnr == stats->bad_seq) {
- /* 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);
- 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 {
- /* duplicate or reordered packet, will be filtered by jitterbuffer. */
- GST_WARNING ("duplicate or reordered packet");
}
- src->stats.octets_received += arrival->payload_len;
- src->stats.bytes_received += arrival->bytes;
- src->stats.packets_received++;
+ 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 += arrival->payload_len;
- /* the source that sent the packet must be a sender */
- src->is_sender = TRUE;
- src->validated = TRUE;
-
- do_bitrate_estimation (src, arrival->running_time, &src->bytes_received);
+ src->bytes_received += pinfo->payload_len;
- GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
+ GST_LOG ("seq %u, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
seqnr, src->stats.packets_received, src->stats.octets_received);
- /* calculate jitter for the stats */
- calculate_jitter (src, buffer, arrival);
-
- /* we're ready to push the RTP packet now */
- result = push_packet (src, buffer);
-
-done:
- return result;
+ return TRUE;
/* ERRORS */
+done:
+ {
+ return FALSE;
+ }
bad_sequence:
{
- GST_WARNING ("unacceptable seqnum received");
- gst_buffer_unref (buffer);
- return GST_FLOW_OK;
+ 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->probation = RTP_DEFAULT_PROBATION;
+ 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;
- gst_buffer_unref (buffer);
- return GST_FLOW_OK;
+ return FALSE;
}
}
/**
- * rtp_source_process_bye:
+ * 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
*
- * Notify @src that a BYE packet has been received. This will make the source
- * inactive.
+ * 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_process_bye (RTPSource * src, const gchar * reason)
+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 received_bye */
+ /* copy the reason and mark as bye */
g_free (src->bye_reason);
src->bye_reason = g_strdup (reason);
- src->received_bye = TRUE;
-}
-
-static gboolean
-set_ssrc (GstBuffer ** buffer, guint idx, RTPSource * src)
-{
- GstRTPBuffer rtp = { NULL };
-
- *buffer = gst_buffer_make_writable (*buffer);
- gst_rtp_buffer_map (*buffer, GST_MAP_WRITE, &rtp);
- gst_rtp_buffer_set_ssrc (&rtp, src->ssrc);
- gst_rtp_buffer_unmap (&rtp);
- return TRUE;
+ src->marked_bye = TRUE;
}
/**
* rtp_source_send_rtp:
* @src: an #RTPSource
- * @data: an RTP buffer or a list of RTP buffers
- * @is_list: if @data is a buffer or list
- * @running_time: the running time of @data
+ * @pinfo: an #RTPPacketInfo
*
- * Send @data (an RTP buffer or list of buffers) originating from @src.
- * This will make @src a sender. This function takes ownership of @data and
+ * 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, gpointer data, gboolean is_list,
- GstClockTime running_time)
+rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
{
GstFlowReturn result;
- guint len;
+ GstClockTime running_time;
guint32 rtptime;
guint64 ext_rtptime;
guint64 rt_diff, rtp_diff;
- GstBufferList *list = NULL;
- GstBuffer *buffer = NULL;
- guint packets;
- guint32 ssrc;
- GstRTPBuffer rtp = { NULL };
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
- g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
-
- if (is_list) {
- list = GST_BUFFER_LIST_CAST (data);
-
- /* We can grab the caps from the first group, since all
- * groups of a buffer list have same caps. */
- buffer = gst_buffer_list_get (list, 0);
- if (!buffer)
- goto no_buffer;
- } else {
- buffer = GST_BUFFER_CAST (data);
- }
/* we are a sender now */
src->is_sender = TRUE;
- if (is_list) {
- gint i;
+ /* we are also a receiver of our packets */
+ if (!update_receiver_stats (src, pinfo, FALSE))
+ return GST_FLOW_OK;
- /* Each group makes up a network packet. */
- packets = gst_buffer_list_length (list);
- for (i = 0, len = 0; i < packets; i++) {
- gst_rtp_buffer_map (gst_buffer_list_get (list, i), GST_MAP_READ, &rtp);
- len += gst_rtp_buffer_get_payload_len (&rtp);
- gst_rtp_buffer_unmap (&rtp);
- }
- /* subsequent info taken from first list member */
- gst_rtp_buffer_map (gst_buffer_list_get (list, 0), GST_MAP_READ, &rtp);
- } else {
- packets = 1;
- gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
- len = gst_rtp_buffer_get_payload_len (&rtp);
+ 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 += packets;
- src->stats.octets_sent += len;
- src->bytes_sent += len;
+ 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 = gst_rtp_buffer_get_timestamp (&rtp);
+ rtptime = pinfo->rtptime;
+
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
src->last_rtptime = ext_rtptime;
/* push packet */
- if (!src->callbacks.push_rtp) {
- gst_rtp_buffer_unmap (&rtp);
+ if (!src->callbacks.push_rtp)
goto no_callback;
- }
-
- ssrc = gst_rtp_buffer_get_ssrc (&rtp);
- gst_rtp_buffer_unmap (&rtp);
-
- if (ssrc != src->ssrc) {
- /* the SSRC of the packet is not correct, make a writable buffer and
- * update the SSRC. This could involve a complete copy of the packet when
- * it is not writable. Usually the payloader will use caps negotiation to
- * get the correct SSRC from the session manager before pushing anything. */
- /* FIXME, we don't want to warn yet because we can't inform any payloader
- * of the changes SSRC yet because we don't implement pad-alloc. */
- GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
- src->ssrc);
-
- if (is_list) {
- list = gst_buffer_list_make_writable (list);
- gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
- } else {
- set_ssrc (&buffer, 0, src);
- }
- }
- GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
- src->stats.packets_sent);
+ GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT,
+ pinfo->is_list ? "list" : "packet", src->stats.packets_sent);
- result = src->callbacks.push_rtp (src, data, src->user_data);
+ result = src->callbacks.push_rtp (src, pinfo->data, src->user_data);
+ pinfo->data = NULL;
return result;
/* ERRORS */
-no_buffer:
- {
- GST_WARNING ("no buffers in buffer list");
- gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
- return GST_FLOW_OK;
- }
no_callback:
{
GST_WARNING ("no callback installed, dropping packet");
- gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
return GST_FLOW_OK;
}
}
* rtp_source_process_sr:
* @src: an #RTPSource
* @time: time of packet arrival
- * @ntptime: the NTP time in 32.32 fixed point
- * @rtptime: the RTP time
+ * @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 octect count
+ * @octet_count: the octet count
*
* Update the sender report in @src.
*/
* @src: an #RTPSource
* @ntpnstime: the current time in nanoseconds since 1970
* @fractionlost: fraction lost since last SR/RR
- * @packetslost: the cumululative number of packets lost
+ * @packetslost: the cumulative number of packets lost
* @exthighestseq: the extended last sequence number received
- * @jitter: the interarrival jitter
- * @lsr: the last SR packet from this source
- * @dlsr: the delay since last SR packet
+ * @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.
*/
* 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 32.32 fixed point
- * @rtptime: the RTP time corresponding to @ntptime
+ * @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 octect 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
+ * 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.
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
* 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) {
- GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
- GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
} else {
diff = -diff;
- GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
- GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
}
} else {
- GST_WARNING ("no clock-rate, cannot interpolate rtp time");
+ 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 */
* @src: an #RTPSource
* @time: the current time of the system clock
* @fractionlost: fraction lost since last SR/RR
- * @packetslost: the cumululative number of packets lost
+ * @packetslost: the cumulative number of packets lost
* @exthighestseq: the extended last sequence number received
- * @jitter: the interarrival jitter
- * @lsr: the last SR packet from this source
- * @dlsr: the delay since last SR packet
+ * @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.
*
* rtp_source_get_last_sr:
* @src: an #RTPSource
* @time: time of packet arrival
- * @ntptime: the NTP time in 32.32 fixed point
- * @rtptime: the RTP time
+ * @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 octect count
+ * @octet_count: the octet count
*
* Get the values of the last sender report as set with rtp_source_process_sr().
*
* rtp_source_get_last_rb:
* @src: an #RTPSource
* @fractionlost: fraction lost since last SR/RR
- * @packetslost: the cumululative number of packets lost
+ * @packetslost: the cumulative number of packets lost
* @exthighestseq: the extended last sequence number received
- * @jitter: the interarrival jitter
- * @lsr: the last SR packet from this source
- * @dlsr: the delay since last SR packet
- * @round_trip: the round trip time
+ * @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().
*
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
rtp_source_find_conflicting_address (RTPSource * src, GSocketAddress * address,
GstClockTime time)
{
- GList *item;
-
- for (item = g_list_first (src->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;
+ return find_conflicting_address (src->conflicting_addresses, address, time);
}
/**
rtp_source_add_conflicting_address (RTPSource * src,
GSocketAddress * address, GstClockTime time)
{
- RTPConflictingAddress *new_conflict;
-
- new_conflict = g_new0 (RTPConflictingAddress, 1);
-
- new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address));
- new_conflict->time = time;
-
- src->conflicting_addresses = g_list_prepend (src->conflicting_addresses,
- new_conflict);
+ src->conflicting_addresses =
+ add_conflicting_address (src->conflicting_addresses, address, time);
}
/**
* rtp_source_timeout:
* @src: The #RTPSource
* @current_time: The current time
- * @collision_timeout: The amount of time after which a collision is timed out
* @feedback_retention_window: The running time before which retained feedback
* packets have to be discarded
*
*/
void
rtp_source_timeout (RTPSource * src, GstClockTime current_time,
- GstClockTime collision_timeout, GstClockTime feedback_retention_window)
+ GstClockTime running_time, GstClockTime feedback_retention_window)
{
- GList *item;
GstRTCPPacket *pkt;
+ GstClockTime max_pts_window;
+ guint pruned = 0;
- item = g_list_first (src->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;
+ src->conflicting_addresses =
+ timeout_conflicting_addresses (src->conflicting_addresses, current_time);
- src->conflicting_addresses =
- g_list_delete_link (src->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);
- g_object_unref (known_conflict->address);
- g_free (known_conflict);
- }
- item = next_item;
+ 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_tail (src->retained_feedback)) &&
- GST_BUFFER_TIMESTAMP (pkt) < feedback_retention_window)
- gst_buffer_unref (g_queue_pop_tail (src->retained_feedback));
+ 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
const GstBuffer *bufa = a;
const GstBuffer *bufb = b;
- return GST_BUFFER_TIMESTAMP (bufa) - GST_BUFFER_TIMESTAMP (bufb);
+ 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
{
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_TIMESTAMP (buffer) = running_time;
+ 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
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);
+ }
+}
projects / platform / upstream / gst-plugins-good.git / blobdiff