diff options
Diffstat (limited to 'gst/rtpmanager/rtpsource.c')
| -rw-r--r-- | gst/rtpmanager/rtpsource.c | 1625 |
1 files changed, 0 insertions, 1625 deletions
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c deleted file mode 100644 index 28fa23ef..00000000 --- a/gst/rtpmanager/rtpsource.c +++ /dev/null @@ -1,1625 +0,0 @@ -/* GStreamer - * Copyright (C) <2007> Wim Taymans <wim.taymans@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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, 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 - -enum -{ - PROP_0, - PROP_SSRC, - PROP_IS_CSRC, - PROP_IS_VALIDATED, - PROP_IS_SENDER, - PROP_SDES, - PROP_STATS, - PROP_LAST -}; - -/* 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 the - * following fields: - * - * 'cname' G_TYPE_STRING : The canonical name - * '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 - */ - 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 - * - * The statistics of the source. This property returns a GstStructure with - * name application/x-rtp-source-stats with the following fields: - * - */ - 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)); - - 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->received_bye = FALSE; - - src->stats.cycles = -1; - src->stats.jitter = 0; - src->stats.transit = -1; - src->stats.curr_sr = 0; - src->stats.curr_rr = 0; -} - -static void -rtp_source_init (RTPSource * src) -{ - /* sources are initialy 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->payload = -1; - src->clock_rate = -1; - src->packets = g_queue_new (); - src->seqnum_base = -1; - src->last_rtptime = -1; - - rtp_source_reset (src); -} - -static void -rtp_source_finalize (GObject * object) -{ - RTPSource *src; - GstBuffer *buffer; - gint i; - - src = RTP_SOURCE_CAST (object); - - while ((buffer = g_queue_pop_head (src->packets))) - gst_buffer_unref (buffer); - g_queue_free (src->packets); - - for (i = 0; i < 9; i++) - g_free (src->sdes[i]); - - g_free (src->bye_reason); - - gst_caps_replace (&src->caps, NULL); - - 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[GST_NETADDRESS_MAX_LEN]; - - /* 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->received_bye, - "is-csrc", G_TYPE_BOOLEAN, src->is_csrc, - "is-sender", G_TYPE_BOOLEAN, is_sender, NULL); - - /* add address and port */ - if (src->have_rtp_from) { - gst_netaddress_to_string (&src->rtp_from, address_str, - sizeof (address_str)); - gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL); - } - if (src->have_rtcp_from) { - gst_netaddress_to_string (&src->rtcp_from, address_str, - sizeof (address_str)); - gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL); - } - - if (internal) { - /* our internal source */ - if (is_sender) { - /* if we are sending, report about how much we sent, other sources will - * have a RB with info on reception. */ - gst_structure_set (s, - "octets-sent", G_TYPE_UINT64, src->stats.octets_sent, - "packets-sent", G_TYPE_UINT64, src->stats.packets_sent, - "bitrate", G_TYPE_UINT64, src->bitrate, NULL); - } else { - /* if we are not sending we have nothing more to report */ - } - } else { - gboolean have_rb; - guint8 fractionlost = 0; - gint32 packetslost = 0; - guint32 exthighestseq = 0; - guint32 jitter = 0; - guint32 lsr = 0; - guint32 dlsr = 0; - guint32 round_trip = 0; - - /* other sources */ - if (is_sender) { - gboolean have_sr; - GstClockTime time = 0; - guint64 ntptime = 0; - guint32 rtptime = 0; - guint32 packet_count = 0; - guint32 octet_count = 0; - - /* this source is sending to us, get the last SR. */ - have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime, - &packet_count, &octet_count); - gst_structure_set (s, - "octets-received", G_TYPE_UINT64, src->stats.octets_received, - "packets-received", G_TYPE_UINT64, src->stats.packets_received, - "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); - } - /* we might be sending to this SSRC so we report about how it is - * receiving our data */ - have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost, - &exthighestseq, &jitter, &lsr, &dlsr, &round_trip); - - gst_structure_set (s, - "have-rb", G_TYPE_BOOLEAN, have_rb, - "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost, - "rb-packetslost", G_TYPE_INT, (gint) packetslost, - "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq, - "rb-jitter", G_TYPE_UINT, (guint) jitter, - "rb-lsr", G_TYPE_UINT, (guint) lsr, - "rb-dlsr", G_TYPE_UINT, (guint) dlsr, - "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL); - } - - return s; -} - -/** - * rtp_source_get_sdes_struct: - * @src: an #RTSPSource - * - * Get the SDES data as a GstStructure - * - * Returns: a GstStructure with SDES items for @src. - */ -GstStructure * -rtp_source_get_sdes_struct (RTPSource * src) -{ - GstStructure *s; - gchar *str; - - s = gst_structure_new ("application/x-rtp-source-sdes", - "ssrc", G_TYPE_UINT, (guint) src->ssrc, NULL); - - if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_CNAME))) { - gst_structure_set (s, "cname", G_TYPE_STRING, str, NULL); - g_free (str); - } - if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NAME))) { - gst_structure_set (s, "name", G_TYPE_STRING, str, NULL); - g_free (str); - } - if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_EMAIL))) { - gst_structure_set (s, "email", G_TYPE_STRING, str, NULL); - g_free (str); - } - if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_PHONE))) { - gst_structure_set (s, "phone", G_TYPE_STRING, str, NULL); - g_free (str); - } - if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_LOC))) { - gst_structure_set (s, "location", G_TYPE_STRING, str, NULL); - g_free (str); - } - if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_TOOL))) { - gst_structure_set (s, "tool", G_TYPE_STRING, str, NULL); - g_free (str); - } - if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NOTE))) { - gst_structure_set (s, "note", G_TYPE_STRING, str, NULL); - g_free (str); - } - return s; -} - -/** - * rtp_source_set_sdes_struct: - * @src: an #RTSPSource - * @sdes: a #GstStructure with SDES info - * - * Set the SDES items from @sdes. - */ -void -rtp_source_set_sdes_struct (RTPSource * src, const GstStructure * sdes) -{ - const gchar *str; - - if (!gst_structure_has_name (sdes, "application/x-rtp-source-sdes")) - return; - - if ((str = gst_structure_get_string (sdes, "cname"))) { - rtp_source_set_sdes_string (src, GST_RTCP_SDES_CNAME, str); - } - if ((str = gst_structure_get_string (sdes, "name"))) { - rtp_source_set_sdes_string (src, GST_RTCP_SDES_NAME, str); - } - if ((str = gst_structure_get_string (sdes, "email"))) { - rtp_source_set_sdes_string (src, GST_RTCP_SDES_EMAIL, str); - } - if ((str = gst_structure_get_string (sdes, "phone"))) { - rtp_source_set_sdes_string (src, GST_RTCP_SDES_PHONE, str); - } - if ((str = gst_structure_get_string (sdes, "location"))) { - rtp_source_set_sdes_string (src, GST_RTCP_SDES_LOC, str); - } - if ((str = gst_structure_get_string (sdes, "tool"))) { - rtp_source_set_sdes_string (src, GST_RTCP_SDES_TOOL, str); - } - if ((str = gst_structure_get_string (sdes, "note"))) { - rtp_source_set_sdes_string (src, GST_RTCP_SDES_NOTE, str); - } -} - -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; - 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_take_boxed (value, rtp_source_get_sdes_struct (src)); - break; - case PROP_STATS: - g_value_take_boxed (value, rtp_source_create_stats (src)); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -/** - * rtp_source_new: - * @ssrc: an SSRC - * - * Create a #RTPSource with @ssrc. - * - * Returns: a new #RTPSource. Use g_object_unref() after usage. - */ -RTPSource * -rtp_source_new (guint32 ssrc) -{ - RTPSource *src; - - src = g_object_new (RTP_TYPE_SOURCE, NULL); - src->ssrc = ssrc; - - return src; -} - -/** - * rtp_source_set_callbacks: - * @src: an #RTPSource - * @cb: callback functions - * @user_data: user data - * - * Set the callbacks for the source. - */ -void -rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb, - gpointer user_data) -{ - g_return_if_fail (RTP_IS_SOURCE (src)); - - src->callbacks.push_rtp = cb->push_rtp; - src->callbacks.clock_rate = cb->clock_rate; - src->user_data = user_data; -} - -/** - * rtp_source_get_ssrc: - * @src: an #RTPSource - * - * Get the SSRC of @source. - * - * Returns: the SSRC of src. - */ -guint32 -rtp_source_get_ssrc (RTPSource * src) -{ - guint32 result; - - g_return_val_if_fail (RTP_IS_SOURCE (src), 0); - - result = src->ssrc; - - return result; -} - -/** - * rtp_source_set_as_csrc: - * @src: an #RTPSource - * - * Configure @src as a CSRC, this will also validate @src. - */ -void -rtp_source_set_as_csrc (RTPSource * src) -{ - g_return_if_fail (RTP_IS_SOURCE (src)); - - src->validated = TRUE; - src->is_csrc = TRUE; -} - -/** - * rtp_source_is_as_csrc: - * @src: an #RTPSource - * - * Check if @src is a contributing source. - * - * Returns: %TRUE if @src is acting as a contributing source. - */ -gboolean -rtp_source_is_as_csrc (RTPSource * src) -{ - gboolean result; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - result = src->is_csrc; - - return result; -} - -/** - * rtp_source_is_active: - * @src: an #RTPSource - * - * Check if @src is an active source. A source is active if it has been - * validated and has not yet received a BYE packet - * - * Returns: %TRUE if @src is an qactive source. - */ -gboolean -rtp_source_is_active (RTPSource * src) -{ - gboolean result; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - result = RTP_SOURCE_IS_ACTIVE (src); - - return result; -} - -/** - * rtp_source_is_validated: - * @src: an #RTPSource - * - * Check if @src is a validated source. - * - * Returns: %TRUE if @src is a validated source. - */ -gboolean -rtp_source_is_validated (RTPSource * src) -{ - gboolean result; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - result = src->validated; - - return result; -} - -/** - * rtp_source_is_sender: - * @src: an #RTPSource - * - * Check if @src is a sending source. - * - * Returns: %TRUE if @src is a sending source. - */ -gboolean -rtp_source_is_sender (RTPSource * src) -{ - gboolean result; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - result = RTP_SOURCE_IS_SENDER (src); - - return result; -} - -/** - * rtp_source_received_bye: - * @src: an #RTPSource - * - * Check if @src has receoved a BYE packet. - * - * Returns: %TRUE if @src has received a BYE packet. - */ -gboolean -rtp_source_received_bye (RTPSource * src) -{ - gboolean result; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - result = src->received_bye; - - 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(). - * - * 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. - */ -gchar * -rtp_source_get_bye_reason (RTPSource * src) -{ - gchar *result; - - g_return_val_if_fail (RTP_IS_SOURCE (src), NULL); - - result = g_strdup (src->bye_reason); - - return result; -} - -/** - * rtp_source_update_caps: - * @src: an #RTPSource - * @caps: a #GstCaps - * - * Parse @caps and store all relevant information in @source. - */ -void -rtp_source_update_caps (RTPSource * src, GstCaps * caps) -{ - GstStructure *s; - guint val; - gint ival; - - /* nothing changed, return */ - if (caps == NULL || src->caps == caps) - return; - - s = gst_caps_get_structure (caps, 0); - - if (gst_structure_get_int (s, "payload", &ival)) - src->payload = ival; - else - src->payload = -1; - GST_DEBUG ("got payload %d", 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, "seqnum-base", &val)) - src->seqnum_base = val; - else - src->seqnum_base = -1; - - GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base); - - gst_caps_replace (&src->caps, caps); -} - -/** - * rtp_source_set_sdes: - * @src: an #RTPSource - * @type: the type of the SDES item - * @data: the SDES data - * @len: the SDES length - * - * 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 (RTPSource * src, GstRTCPSDESType type, - const guint8 * data, guint len) -{ - guint8 *old; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - if (type < 0 || type > GST_RTCP_SDES_PRIV) - return FALSE; - - old = src->sdes[type]; - - /* lengths are the same, check if the data is the same */ - if ((src->sdes_len[type] == len)) - if (data != NULL && old != NULL && (memcmp (old, data, len) == 0)) - return FALSE; - - /* NULL data, make sure we store 0 length or if no length is given, - * take strlen */ - if (data == NULL) - len = 0; - - g_free (src->sdes[type]); - src->sdes[type] = g_memdup (data, len); - src->sdes_len[type] = len; - - return TRUE; -} - -/** - * 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. This function is similar to - * rtp_source_set_sdes() but takes a null-terminated string for convenience. - * - * 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) -{ - guint len; - gboolean result; - - if (data) - len = strlen (data); - else - len = 0; - - result = rtp_source_set_sdes (src, type, (guint8 *) data, len); - - return result; -} - -/** - * rtp_source_get_sdes: - * @src: an #RTPSource - * @type: the type of the SDES item - * @data: location to store the SDES data or NULL - * @len: location to store the SDES length or NULL - * - * Get the SDES item of @type from @src. Note that @data does not always point - * to a null-terminated string, use rtp_source_get_sdes_string() to retrieve a - * null-terminated string instead. - * - * @data remains valid until the next call to rtp_source_set_sdes(). - * - * Returns: %TRUE if @type was valid and @data and @len contain valid - * data. @data can be NULL when the item was unset. - */ -gboolean -rtp_source_get_sdes (RTPSource * src, GstRTCPSDESType type, guint8 ** data, - guint * len) -{ - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - if (type < 0 || type > GST_RTCP_SDES_PRIV) - return FALSE; - - if (data) - *data = src->sdes[type]; - if (len) - *len = src->sdes_len[type]; - - 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; - - g_return_val_if_fail (RTP_IS_SOURCE (src), NULL); - - if (type < 0 || type > GST_RTCP_SDES_PRIV) - return NULL; - - result = g_strndup ((const gchar *) src->sdes[type], src->sdes_len[type]); - - return result; -} - -/** - * 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, GstNetAddress * address) -{ - g_return_if_fail (RTP_IS_SOURCE (src)); - - src->have_rtp_from = TRUE; - memcpy (&src->rtp_from, address, sizeof (GstNetAddress)); -} - -/** - * 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, GstNetAddress * address) -{ - g_return_if_fail (RTP_IS_SOURCE (src)); - - src->have_rtcp_from = TRUE; - memcpy (&src->rtcp_from, address, sizeof (GstNetAddress)); -} - -static GstFlowReturn -push_packet (RTPSource * src, GstBuffer * buffer) -{ - GstFlowReturn ret = GST_FLOW_OK; - - /* push queued packets first if any */ - while (!g_queue_is_empty (src->packets)) { - GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets)); - - GST_LOG ("pushing queued packet"); - if (src->callbacks.push_rtp) - src->callbacks.push_rtp (src, buffer, src->user_data); - else - gst_buffer_unref (buffer); - } - GST_LOG ("pushing new packet"); - /* push packet */ - if (src->callbacks.push_rtp) - ret = src->callbacks.push_rtp (src, buffer, src->user_data); - else - gst_buffer_unref (buffer); - - return ret; -} - -static gint -get_clock_rate (RTPSource * src, guint8 payload) -{ - if (src->payload == -1) { - /* first payload received, nothing was in the caps, lock on to this payload */ - src->payload = payload; - GST_DEBUG ("first payload %d", payload); - } else if (payload != src->payload) { - /* we have a different payload than before, reset the clock-rate */ - GST_DEBUG ("new payload %d", payload); - src->payload = payload; - src->clock_rate = -1; - src->stats.transit = -1; - } - - if (src->clock_rate == -1) { - gint clock_rate = -1; - - if (src->callbacks.clock_rate) - clock_rate = src->callbacks.clock_rate (src, payload, src->user_data); - - GST_DEBUG ("got clock-rate %d", clock_rate); - - src->clock_rate = clock_rate; - } - return src->clock_rate; -} - -/* Jitter is the variation in the delay of received packets in a flow. It is - * measured by comparing the interval when RTP packets were sent to the interval - * at which they were received. For instance, if packet #1 and packet #2 leave - * 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10 - * milliseconds. */ -static void -calculate_jitter (RTPSource * src, GstBuffer * buffer, - RTPArrivalStats * arrival) -{ - guint64 ntpnstime; - guint32 rtparrival, transit, rtptime; - gint32 diff; - gint clock_rate; - guint8 pt; - - /* get arrival time */ - if ((ntpnstime = arrival->ntpnstime) == GST_CLOCK_TIME_NONE) - goto no_time; - - pt = gst_rtp_buffer_get_payload_type (buffer); - - GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt); - - /* get clockrate */ - if ((clock_rate = get_clock_rate (src, pt)) == -1) - goto no_clock_rate; - - rtptime = gst_rtp_buffer_get_timestamp (buffer); - - /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't - * care about the absolute value, just the difference. */ - rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND); - - /* transit time is difference with RTP timestamp */ - transit = rtparrival - rtptime; - - /* get ABS diff with previous transit time */ - if (src->stats.transit != -1) { - if (transit > src->stats.transit) - diff = transit - src->stats.transit; - else - diff = src->stats.transit - transit; - } else - diff = 0; - - src->stats.transit = transit; - - /* update jitter, the value we store is scaled up so we can keep precision. */ - src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4); - - src->stats.prev_rtptime = src->stats.last_rtptime; - src->stats.last_rtptime = rtparrival; - - GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f", - rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0); - - return; - - /* ERRORS */ -no_time: - { - GST_WARNING ("cannot get current time"); - return; - } -no_clock_rate: - { - GST_WARNING ("cannot get clock-rate for pt %d", pt); - return; - } -} - -static void -init_seq (RTPSource * src, guint16 seq) -{ - src->stats.base_seq = seq; - src->stats.max_seq = seq; - src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */ - src->stats.cycles = 0; - src->stats.packets_received = 0; - src->stats.octets_received = 0; - src->stats.bytes_received = 0; - src->stats.prev_received = 0; - src->stats.prev_expected = 0; - - 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) -{ - GstFlowReturn result = GST_FLOW_OK; - guint16 seqnr, udelta; - RTPSourceStats *stats; - - g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR); - g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR); - - stats = &src->stats; - - seqnr = gst_rtp_buffer_get_seq (buffer); - - rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer)); - - if (stats->cycles == -1) { - GST_DEBUG ("received first buffer"); - /* first time we heard of this source */ - init_seq (src, seqnr); - src->stats.max_seq = seqnr - 1; - src->probation = RTP_DEFAULT_PROBATION; - } - - udelta = seqnr - stats->max_seq; - - /* if we are still on probation, check seqnum */ - if (src->probation) { - guint16 expected; - - 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!"); - 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; - } - } else { - GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected); - src->probation = RTP_DEFAULT_PROBATION; - src->stats.max_seq = seqnr; - goto done; - } - } 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 { - /* 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++; - /* the source that sent the packet must be a sender */ - src->is_sender = TRUE; - src->validated = TRUE; - - GST_LOG ("seq %d, 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; - - /* ERRORS */ -bad_sequence: - { - GST_WARNING ("unacceptable seqnum received"); - gst_buffer_unref (buffer); - return GST_FLOW_OK; - } -} - -/** - * rtp_source_process_bye: - * @src: an #RTPSource - * @reason: the reason for leaving - * - * Notify @src that a BYE packet has been received. This will make the source - * inactive. - */ -void -rtp_source_process_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 */ - g_free (src->bye_reason); - src->bye_reason = g_strdup (reason); - src->received_bye = TRUE; -} - -static GstBufferListItem -set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src) -{ - *buffer = gst_buffer_make_writable (*buffer); - gst_rtp_buffer_set_ssrc (*buffer, src->ssrc); - return GST_BUFFER_LIST_SKIP_GROUP; -} - -/** - * 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 - * @ntpnstime: the NTP time when this buffer was captured in nanoseconds. This - * is the buffer timestamp converted to NTP time. - * - * 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 - * 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, - guint64 ntpnstime) -{ - GstFlowReturn result; - guint len; - guint32 rtptime; - guint64 ext_rtptime; - guint64 ntp_diff, rtp_diff; - guint64 elapsed; - GstBufferList *list = NULL; - GstBuffer *buffer = NULL; - guint packets; - guint32 ssrc; - - 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, 0); - if (!buffer) - goto no_buffer; - } else { - buffer = GST_BUFFER_CAST (data); - } - rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer)); - - /* we are a sender now */ - src->is_sender = TRUE; - - if (is_list) { - /* Each group makes up a network packet. */ - packets = gst_buffer_list_n_groups (list); - len = gst_rtp_buffer_list_get_payload_len (list); - } else { - packets = 1; - len = gst_rtp_buffer_get_payload_len (buffer); - } - - /* update stats for the SR */ - src->stats.packets_sent += packets; - src->stats.octets_sent += len; - src->bytes_sent += len; - - if (src->prev_ntpnstime) { - elapsed = ntpnstime - src->prev_ntpnstime; - - if (elapsed > (G_GINT64_CONSTANT (1) << 31)) { - guint64 rate; - - rate = - gst_util_uint64_scale (src->bytes_sent, elapsed, - (G_GINT64_CONSTANT (1) << 29)); - - GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT - ", rate %" G_GUINT64_FORMAT, elapsed, src->bytes_sent, rate); - - if (src->bitrate == 0) - src->bitrate = rate; - else - src->bitrate = ((src->bitrate * 3) + rate) / 4; - - src->prev_ntpnstime = ntpnstime; - src->bytes_sent = 0; - } - } else { - GST_LOG ("Reset bitrate measurement"); - src->prev_ntpnstime = ntpnstime; - src->bitrate = 0; - } - - if (is_list) { - rtptime = gst_rtp_buffer_list_get_timestamp (list); - } else { - rtptime = gst_rtp_buffer_get_timestamp (buffer); - } - ext_rtptime = src->last_rtptime; - ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime); - - GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT, - src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime)); - - if (ext_rtptime > src->last_rtptime) { - rtp_diff = ext_rtptime - src->last_rtptime; - ntp_diff = ntpnstime - src->last_ntpnstime; - - /* 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 NTP %" - GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff)); - } - - /* we keep track of the last received RTP timestamp and the corresponding - * NTP timestamp so that we can use this info when constructing SR reports */ - src->last_rtptime = ext_rtptime; - src->last_ntpnstime = ntpnstime; - - /* push packet */ - if (!src->callbacks.push_rtp) - goto no_callback; - - if (is_list) { - ssrc = gst_rtp_buffer_list_get_ssrc (list); - } else { - ssrc = gst_rtp_buffer_get_ssrc (buffer); - } - - 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, 0, src); - } - } - GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet", - src->stats.packets_sent); - - result = src->callbacks.push_rtp (src, data, src->user_data); - - 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 - * @packet_count: the packet count - * @octet_count: the octect 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; -} - -/** - * rtp_source_process_rb: - * @src: an #RTPSource - * @time: the current time in nanoseconds since 1970 - * @fractionlost: fraction lost since last SR/RR - * @packetslost: the cumululative 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 - * - * Update the report block in @src. - */ -void -rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost, - gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr, - guint32 dlsr) -{ - RTPReceiverReport *curr; - gint curridx; - guint32 ntp, A; - - g_return_if_fail (RTP_IS_SOURCE (src)); - - GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT - ", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x", - src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16, - lsr & 0xffff, dlsr >> 16, dlsr & 0xffff); - - curridx = src->stats.curr_rr ^ 1; - curr = &src->stats.rr[curridx]; - - /* update current */ - curr->is_valid = TRUE; - curr->fractionlost = fractionlost; - curr->packetslost = packetslost; - curr->exthighestseq = exthighestseq; - curr->jitter = jitter; - curr->lsr = lsr; - curr->dlsr = dlsr; - - /* calculate round trip, round the time up */ - ntp = ((gst_rtcp_unix_to_ntp (time) + 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 - * @ntptime: the NTP time in 32.32 fixed point - * @rtptime: the RTP time corresponding to @ntptime - * @packet_count: the packet count - * @octet_count: the octect count - * - * Get new values to put into a new SR report from this source. - * - * Returns: %TRUE on success. - */ -gboolean -rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime, - 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); - - /* use the sync params to interpolate the date->time member to rtptime. We - * use the last sent timestamp and rtptime as reference points. We assume - * that the slope of the rtptime vs timestamp curve is 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_ntpnstime %" GST_TIME_FORMAT ", last_rtptime %" - G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_ntpnstime), t_rtp); - - if (src->clock_rate != -1) { - /* get the diff with the SR time */ - diff = GST_CLOCK_DIFF (src->last_ntpnstime, ntpnstime); - - /* 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 ("ntpnstime %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT, - GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff)); - t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND); - } else { - diff = -diff; - GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT, - GST_TIME_ARGS (ntpnstime), 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"); - } - - /* 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 cumululative 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 - * - * 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 32.32 fixed point - * @rtptime: the RTP time - * @packet_count: the packet count - * @octet_count: the octect count - * - * Get the values of the last sender report as set with rtp_source_process_sr(). - * - * Returns: %TRUE if there was a valid SR report. - */ -gboolean -rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime, - guint32 * rtptime, guint32 * packet_count, guint32 * octet_count) -{ - RTPSenderReport *curr; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - curr = &src->stats.sr[src->stats.curr_sr]; - if (!curr->is_valid) - return FALSE; - - if (ntptime) - *ntptime = curr->ntptime; - if (rtptime) - *rtptime = curr->rtptime; - if (packet_count) - *packet_count = curr->packet_count; - if (octet_count) - *octet_count = curr->octet_count; - if (time) - *time = curr->time; - - return TRUE; -} - -/** - * rtp_source_get_last_rb: - * @src: an #RTPSource - * @fractionlost: fraction lost since last SR/RR - * @packetslost: the cumululative 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 - * - * Get the values of the last RB report set with rtp_source_process_rb(). - * - * Returns: %TRUE if there was a valid SB report. - */ -gboolean -rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost, - gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter, - guint32 * lsr, guint32 * dlsr, guint32 * round_trip) -{ - RTPReceiverReport *curr; - - g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); - - curr = &src->stats.rr[src->stats.curr_rr]; - if (!curr->is_valid) - return FALSE; - - if (fractionlost) - *fractionlost = curr->fractionlost; - if (packetslost) - *packetslost = curr->packetslost; - if (exthighestseq) - *exthighestseq = curr->exthighestseq; - if (jitter) - *jitter = curr->jitter; - if (lsr) - *lsr = curr->lsr; - if (dlsr) - *dlsr = curr->dlsr; - if (round_trip) - *round_trip = curr->round_trip; - - return TRUE; -} |