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gst/rtpmanager/gstrtpbin.c: fix for pad name change

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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.
This commit is contained in:
Wim Taymans 2007-04-25 08:30:48 +00:00
parent 203ed49721
commit 34534179a2
9 changed files with 668 additions and 89 deletions
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40
ChangeLog
View file

@ -1,3 +1,43 @@
2007-04-25 Wim Taymans <wim@fluendo.com>
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.
2007-04-24 Tim-Philipp Müller <tim at centricular dot net>
* gst/y4m/gsty4mencode.c: (gst_y4m_encode_init),

2
gst/rtpmanager/gstrtpbin.c
View file

@ -959,7 +959,7 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* get rtcp_src pad and store */
session->rtcp_src =
gst_element_get_request_pad (session->session, "rtcp_src");
gst_element_get_request_pad (session->session, "send_rtcp_src");
if (session->rtcp_src == NULL)
goto pad_failed;

8
gst/rtpmanager/gstrtpsession.c
View file

@ -305,7 +305,7 @@ rtcp_thread (GstRTPSession * rtpsession)
gdouble timeout;
GstClockTime target;
timeout = rtp_session_get_rtcp_interval (rtpsession->priv->session);
timeout = rtp_session_get_reporting_interval (rtpsession->priv->session);
GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout);
target = gst_clock_get_time (clock);
@ -318,7 +318,7 @@ rtcp_thread (GstRTPSession * rtpsession)
GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
/* make the session manager produce RTCP, we ignore the result. */
rtp_session_produce_rtcp (rtpsession->priv->session);
rtp_session_perform_reporting (rtpsession->priv->session);
GST_RTP_SESSION_LOCK (rtpsession);
gst_clock_id_unref (id);
@ -472,6 +472,8 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
if (rtpsession->send_rtcp_src) {
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
} else {
@ -515,6 +517,8 @@ gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
if (!gst_structure_get_int (caps_struct, "clock-rate", &result))
goto no_clock_rate;
GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
return result;
/* ERRORS */

489
gst/rtpmanager/rtpsession.c
View file

@ -57,6 +57,9 @@ static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
static void
rtp_session_class_init (RTPSessionClass * klass)
{
@ -123,18 +126,35 @@ rtp_session_class_init (RTPSessionClass * klass)
static void
rtp_session_init (RTPSession * sess)
{
sess->lock = g_mutex_new ();
sess->ssrcs =
g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) g_object_unref);
sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
gint i;
/* create an SSRC for this session manager */
sess->source = rtp_session_create_source (sess);
sess->lock = g_mutex_new ();
sess->key = g_random_int ();
sess->mask_idx = 0;
sess->mask = 0;
for (i = 0; i < 32; i++) {
sess->ssrcs[i] =
g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) g_object_unref);
}
sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
rtp_stats_init_defaults (&sess->stats);
/* create an active SSRC for this session manager */
sess->source = rtp_session_create_source (sess);
sess->stats.active_sources++;
/* default UDP header length */
sess->header_len = 28;
sess->mtu = 1400;
/* some default SDES entries */
//sess->cname = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
sess->cname = g_strdup_printf ("foo@%s", g_get_host_name ());
sess->name = g_strdup (g_get_real_name ());
sess->tool = g_strdup ("GStreamer");
GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
}
@ -143,14 +163,20 @@ static void
rtp_session_finalize (GObject * object)
{
RTPSession *sess;
gint i;
sess = RTP_SESSION_CAST (object);
g_mutex_free (sess->lock);
g_hash_table_destroy (sess->ssrcs);
for (i = 0; i < 32; i++)
g_hash_table_destroy (sess->ssrcs[i]);
g_hash_table_destroy (sess->cnames);
g_object_unref (sess->source);
g_free (sess->cname);
g_free (sess->tool);
G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
}
@ -312,6 +338,230 @@ rtp_session_get_rtcp_bandwidth (RTPSession * sess)
return sess->stats.rtcp_bandwidth;
}
/**
* rtp_session_set_cname:
* @sess: an #RTPSession
* @cname: a CNAME for the session
*
* Set the CNAME for the session.
*/
void
rtp_session_set_cname (RTPSession * sess, const gchar * cname)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->cname);
sess->cname = g_strdup (cname);
}
/**
* rtp_session_get_cname:
* @sess: an #RTPSession
*
* Get the currently configured CNAME for the session.
*
* Returns: The CNAME. g_free after usage.
*/
gchar *
rtp_session_get_cname (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->cname);
}
/**
* rtp_session_set_name:
* @sess: an #RTPSession
* @name: a NAME for the session
*
* Set the NAME for the session.
*/
void
rtp_session_set_name (RTPSession * sess, const gchar * name)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->name);
sess->name = g_strdup (name);
}
/**
* rtp_session_get_name:
* @sess: an #RTPSession
*
* Get the currently configured NAME for the session.
*
* Returns: The NAME. g_free after usage.
*/
gchar *
rtp_session_get_name (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->name);
}
/**
* rtp_session_set_email:
* @sess: an #RTPSession
* @email: an EMAIL for the session
*
* Set the EMAIL the session.
*/
void
rtp_session_set_email (RTPSession * sess, const gchar * email)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->email);
sess->email = g_strdup (email);
}
/**
* rtp_session_get_email:
* @sess: an #RTPSession
*
* Get the currently configured EMAIL of the session.
*
* Returns: The EMAIL. g_free after usage.
*/
gchar *
rtp_session_get_email (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->email);
}
/**
* rtp_session_set_phone:
* @sess: an #RTPSession
* @phone: a PHONE for the session
*
* Set the PHONE the session.
*/
void
rtp_session_set_phone (RTPSession * sess, const gchar * phone)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->phone);
sess->phone = g_strdup (phone);
}
/**
* rtp_session_get_location:
* @sess: an #RTPSession
*
* Get the currently configured PHONE of the session.
*
* Returns: The PHONE. g_free after usage.
*/
gchar *
rtp_session_get_phone (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->phone);
}
/**
* rtp_session_set_location:
* @sess: an #RTPSession
* @location: a LOCATION for the session
*
* Set the LOCATION the session.
*/
void
rtp_session_set_location (RTPSession * sess, const gchar * location)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->location);
sess->location = g_strdup (location);
}
/**
* rtp_session_get_location:
* @sess: an #RTPSession
*
* Get the currently configured LOCATION of the session.
*
* Returns: The LOCATION. g_free after usage.
*/
gchar *
rtp_session_get_location (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->location);
}
/**
* rtp_session_set_tool:
* @sess: an #RTPSession
* @tool: a TOOL for the session
*
* Set the TOOL the session.
*/
void
rtp_session_set_tool (RTPSession * sess, const gchar * tool)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->tool);
sess->tool = g_strdup (tool);
}
/**
* rtp_session_get_tool:
* @sess: an #RTPSession
*
* Get the currently configured TOOL of the session.
*
* Returns: The TOOL. g_free after usage.
*/
gchar *
rtp_session_get_tool (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->tool);
}
/**
* rtp_session_set_note:
* @sess: an #RTPSession
* @note: a NOTE for the session
*
* Set the NOTE the session.
*/
void
rtp_session_set_note (RTPSession * sess, const gchar * note)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->note);
sess->note = g_strdup (note);
}
/**
* rtp_session_get_note:
* @sess: an #RTPSession
*
* Get the currently configured NOTE of the session.
*
* Returns: The NOTE. g_free after usage.
*/
gchar *
rtp_session_get_note (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->note);
}
static GstFlowReturn
source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
{
@ -319,6 +569,8 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
if (source == session->source) {
GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
if (session->callbacks.send_rtp)
result =
session->callbacks.send_rtp (session, source, buffer,
@ -371,7 +623,8 @@ obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
{
RTPSource *source;
source = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc));
source =
g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
if (source == NULL) {
/* make new Source in probation and insert */
source = rtp_source_new (ssrc);
@ -392,7 +645,8 @@ obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
/* configure a callback on the source */
rtp_source_set_callbacks (source, &callbacks, sess);
g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (ssrc), source);
g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
source);
/* we have one more source now */
sess->total_sources++;
@ -426,9 +680,12 @@ rtp_session_add_source (RTPSession * sess, RTPSource * src)
g_return_val_if_fail (src != NULL, FALSE);
RTP_SESSION_LOCK (sess);
find = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (src->ssrc));
find =
g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (src->ssrc));
if (find == NULL) {
g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (src->ssrc), src);
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (src->ssrc), src);
/* we have one more source now */
sess->total_sources++;
result = TRUE;
@ -501,7 +758,8 @@ rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
RTP_SESSION_LOCK (sess);
result = g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc));
result =
g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
if (result)
g_object_ref (result);
RTP_SESSION_UNLOCK (sess);
@ -556,11 +814,13 @@ rtp_session_create_source (RTPSession * sess)
ssrc = g_random_int ();
/* see if it exists in the session, we're done if it doesn't */
if (g_hash_table_lookup (sess->ssrcs, GINT_TO_POINTER (ssrc)) == NULL)
if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (ssrc)) == NULL)
break;
}
source = rtp_source_new (ssrc);
g_hash_table_insert (sess->ssrcs, GINT_TO_POINTER (ssrc), source);
g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
source);
/* we have one more source now */
sess->total_sources++;
RTP_SESSION_UNLOCK (sess);
@ -633,6 +893,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
prevsender = RTP_SOURCE_IS_SENDER (source);
prevactive = RTP_SOURCE_IS_ACTIVE (source);
gst_buffer_ref (buffer);
/* let source process the packet */
result = rtp_source_process_rtp (source, buffer, &arrival);
@ -652,10 +914,11 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
if (created)
on_new_ssrc (sess, source);
/* for validated sources, we add the CSRCs as well */
if (source->validated) {
guint8 i, count;
gboolean created;
/* for validated sources, we add the CSRCs as well */
count = gst_rtp_buffer_get_csrc_count (buffer);
for (i = 0; i < count; i++) {
@ -675,6 +938,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
}
}
}
gst_buffer_unref (buffer);
RTP_SESSION_UNLOCK (sess);
return result;
@ -704,17 +969,27 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
guint64 ntptime;
guint count, i;
RTPSource *source;
gboolean created;
gboolean created, prevsender;
gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
&packet_count, &octet_count);
GST_DEBUG ("got SR packet: SSRC %08x", senderssrc);
RTP_SESSION_LOCK (sess);
source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
prevsender = RTP_SOURCE_IS_SENDER (source);
/* first update the source */
rtp_source_process_sr (source, ntptime, rtptime, packet_count, octet_count);
if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
sess->stats.sender_sources++;
GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
sess->stats.sender_sources);
}
if (created)
on_new_ssrc (sess, source);
@ -785,36 +1060,36 @@ static void
rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival)
{
guint chunks, i, j;
gboolean more_chunks, more_items;
guint items, i, j;
gboolean more_items, more_entries;
chunks = gst_rtcp_packet_sdes_get_chunk_count (packet);
GST_DEBUG ("got SDES packet with %d chunks", chunks);
items = gst_rtcp_packet_sdes_get_item_count (packet);
GST_DEBUG ("got SDES packet with %d items", items);
more_chunks = gst_rtcp_packet_sdes_first_chunk (packet);
more_items = gst_rtcp_packet_sdes_first_item (packet);
i = 0;
while (more_chunks) {
while (more_items) {
guint32 ssrc;
ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
GST_DEBUG ("chunk %d, SSRC %08x", i, ssrc);
GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
more_items = gst_rtcp_packet_sdes_first_item (packet);
more_entries = gst_rtcp_packet_sdes_first_entry (packet);
j = 0;
while (more_items) {
while (more_entries) {
GstRTCPSDESType type;
guint8 len;
gchar *data;
guint8 *data;
gst_rtcp_packet_sdes_get_item (packet, &type, &len, &data);
gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
GST_DEBUG ("item %d, type %d, len %d, data %s", j, type, len, data);
GST_DEBUG ("entry %d, type %d, len %d, data %s", j, type, len, data);
more_items = gst_rtcp_packet_sdes_next_item (packet);
more_entries = gst_rtcp_packet_sdes_next_entry (packet);
j++;
}
more_chunks = gst_rtcp_packet_sdes_next_chunk (packet);
more_items = gst_rtcp_packet_sdes_next_item (packet);
i++;
}
}
@ -906,6 +1181,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
GST_DEBUG ("received RTCP packet");
/* get packet size including header overhead */
RTP_SESSION_LOCK (sess);
size = GST_BUFFER_SIZE (buffer) + sess->header_len;
/* update average RTCP packet size */
@ -914,6 +1190,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
else
sess->stats.avg_rtcp_packet_size =
(size + (15 * sess->stats.avg_rtcp_packet_size)) >> 4;
RTP_SESSION_UNLOCK (sess);
/* start processing the compound packet */
more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
@ -972,6 +1249,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
RTP_SESSION_LOCK (sess);
source = sess->source;
prevsender = RTP_SOURCE_IS_SENDER (source);
@ -981,20 +1259,22 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
sess->stats.sender_sources++;
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_get_rtcp_interval:
* rtp_session_get_reporting_interval:
* @sess: an #RTPSession
*
* Get the interval for sending out the next RTCP packet
* Get the interval for sending out the next RTCP packet and doing general
* maintenance tasks.
*
* Returns: an interval in seconds.
*/
gdouble
rtp_session_get_rtcp_interval (RTPSession * sess)
rtp_session_get_reporting_interval (RTPSession * sess)
{
gdouble result;
@ -1008,8 +1288,112 @@ rtp_session_get_rtcp_interval (RTPSession * sess)
return result;
}
typedef struct
{
RTPSession *sess;
GstBuffer *rtcp;
GstRTCPPacket packet;
} ReportData;
static void
session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
{
RTPSession *sess = data->sess;
RTPSource *own = sess->source;
GstRTCPPacket *packet = &data->packet;
/* create a new buffer if needed */
if (data->rtcp == NULL) {
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
if (RTP_SOURCE_IS_SENDER (own)) {
/* we are a sender, create SR */
GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
/* fill in sender report info */
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
0, 0, own->stats.packets_sent, own->stats.octets_sent);
} else {
/* we are only receiver, create RR */
GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
}
}
if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
/* only report about other sources */
if (source != sess->source) {
RTPSourceStats *stats;
guint32 extended_max, expected;
guint32 expected_interval, received_interval;
guint32 lost, lost_interval, fraction;
stats = &source->stats;
extended_max = (stats->cycles << 16) + stats->max_seq;
expected = extended_max - stats->base_seq + 1;
if (expected > stats->packets_received) {
lost = expected - stats->packets_received;
if (lost > 0x7fffff)
lost = 0x7fffff;
} else {
lost = stats->packets_received - expected;
if (lost > 0x800000)
lost = 0x800000;
else
lost = -lost;
}
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", source->ssrc);
/* we scaled the jitter up for additional precision */
GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
extended_max, stats->jitter >> 4);
/* packet is not yet filled, add report block for this source. */
gst_rtcp_packet_add_rb (packet, source->ssrc, fraction, lost,
extended_max, stats->jitter >> 4, 0, 0);
}
}
}
static void
session_sdes (RTPSession * sess, GstBuffer * buffer)
{
GstRTCPPacket packet;
/* add SDES packet */
gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_SDES, &packet);
gst_rtcp_packet_sdes_add_item (&packet, sess->source->ssrc);
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_CNAME,
strlen (sess->cname), (guint8 *) sess->cname);
/* other SDES items must only be added at regular intervals and only when the
* user requests to since it might be a privacy problem */
#if 0
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
strlen (sess->name), (guint8 *) sess->name);
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
strlen (sess->tool), (guint8 *) sess->tool);
#endif
}
/**
* rtp_session_produce_rtcp:
* rtp_session_perform_reporting:
* @sess: an #RTPSession
*
* Instruct the session manager to generate RTCP packets with current stats.
@ -1019,8 +1403,39 @@ rtp_session_get_rtcp_interval (RTPSession * sess)
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_produce_rtcp (RTPSession * sess)
rtp_session_perform_reporting (RTPSession * sess)
{
/* FIXME: implement me */
return GST_FLOW_NOT_SUPPORTED;
GstFlowReturn result = GST_FLOW_OK;
ReportData data;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
data.sess = sess;
data.rtcp = NULL;
RTP_SESSION_LOCK (sess);
/* loop over all known sources and do something */
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) session_report_blocks, &data);
/* add SDES for this source */
if (data.rtcp) {
session_sdes (sess, data.rtcp);
sess->stats.sent_rtcp = TRUE;
}
RTP_SESSION_UNLOCK (sess);
/* push out the RTCP packet */
if (data.rtcp) {
/* close the RTCP packet */
gst_rtcp_buffer_end (data.rtcp);
if (sess->callbacks.send_rtcp)
result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
sess->user_data);
else
gst_buffer_unref (data.rtcp);
}
return result;
}

36
gst/rtpmanager/rtpsession.h
View file

@ -143,9 +143,24 @@ struct _RTPSession {
GMutex *lock;
guint header_len;
guint mtu;
RTPSource *source;
GHashTable *ssrcs;
/* info for creating reports */
gchar *cname;
gchar *name;
gchar *email;
gchar *phone;
gchar *location;
gchar *tool;
gchar *note;
/* for sender/receiver counting */
guint32 key;
guint32 mask_idx;
guint32 mask;
GHashTable *ssrcs[32];
GHashTable *cnames;
guint total_sources;
@ -184,6 +199,21 @@ gdouble rtp_session_get_bandwidth (RTPSession *sess);
void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction);
gdouble rtp_session_get_rtcp_fraction (RTPSession *sess);
void rtp_session_set_cname (RTPSession *sess, const gchar *cname);
gchar* rtp_session_get_cname (RTPSession *sess);
void rtp_session_set_name (RTPSession *sess, const gchar *name);
gchar* rtp_session_get_name (RTPSession *sess);
void rtp_session_set_email (RTPSession *sess, const gchar *email);
gchar* rtp_session_get_email (RTPSession *sess);
void rtp_session_set_phone (RTPSession *sess, const gchar *phone);
gchar* rtp_session_get_phone (RTPSession *sess);
void rtp_session_set_location (RTPSession *sess, const gchar *location);
gchar* rtp_session_get_location (RTPSession *sess);
void rtp_session_set_tool (RTPSession *sess, const gchar *tool);
gchar* rtp_session_get_tool (RTPSession *sess);
void rtp_session_set_note (RTPSession *sess, const gchar *note);
gchar* rtp_session_get_note (RTPSession *sess);
/* handling sources */
gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src);
gint rtp_session_get_num_sources (RTPSession *sess);
@ -200,7 +230,7 @@ GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer
GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer);
/* get interval for next RTCP interval */
gdouble rtp_session_get_rtcp_interval (RTPSession *sess);
GstFlowReturn rtp_session_produce_rtcp (RTPSession *sess);
gdouble rtp_session_get_reporting_interval (RTPSession *sess);
GstFlowReturn rtp_session_perform_reporting (RTPSession *sess);
#endif /* __RTP_SESSION_H__ */

104
gst/rtpmanager/rtpsource.c
View file

@ -70,6 +70,7 @@ rtp_source_init (RTPSource * src)
src->clock_rate = -1;
src->packets = g_queue_new ();
src->stats.cycles = -1;
src->stats.jitter = 0;
src->stats.transit = -1;
src->stats.curr_sr = 0;
@ -279,6 +280,20 @@ no_clock_rate:
}
}
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;
}
/**
* rtp_source_process_rtp:
* @src: an #RTPSource
@ -293,31 +308,42 @@ 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);
if (stats->cycles == -1) {
GST_DEBUG ("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 seqnr, expected;
guint16 expected;
expected = src->stats.max_seqnr + 1;
expected = src->stats.max_seq + 1;
/* when in probation, we require consecutive seqnums */
seqnr = gst_rtp_buffer_get_seq (buffer);
if (seqnr == expected) {
/* expected packet */
src->probation--;
src->stats.max_seqnr = seqnr;
GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
src->probation--;
src->stats.max_seq = seqnr;
if (src->probation == 0) {
GST_DEBUG ("probation done!", src->probation);
init_seq (src, seqnr);
} else {
GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
src->probation = RTP_DEFAULT_PROBATION;
src->stats.max_seqnr = seqnr;
}
}
if (src->probation) {
GstBuffer *q;
GST_DEBUG ("probation %d: queue buffer", src->probation);
@ -328,23 +354,62 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
q = g_queue_pop_head (src->packets);
gst_object_unref (q);
}
goto done;
}
} else {
/* we are not in probation */
src->stats.octetsreceived += arrival->payload_len;
src->stats.bytesreceived += arrival->bytes;
src->stats.packetsreceived++;
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++;
}
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. */
}
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_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
src->stats.packetsreceived, src->stats.octetsreceived);
src->stats.packets_received, src->stats.octets_received);
/* calculate jitter */
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");
return GST_FLOW_OK;
}
}
/**
@ -424,6 +489,9 @@ rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
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;

29
gst/rtpmanager/rtpsource.h
View file

@ -31,6 +31,10 @@
#define RTP_NO_PROBATION 0
#define RTP_DEFAULT_PROBATION 2
#define RTP_SEQ_MOD (1 << 16)
#define RTP_MAX_DROPOUT 3000
#define RTP_MAX_MISORDER 100
typedef struct _RTPSource RTPSource;
typedef struct _RTPSourceClass RTPSourceClass;
@ -69,7 +73,8 @@ typedef struct _RTPSourceClass RTPSourceClass;
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, gpointer user_data);
typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer,
gpointer user_data);
/**
* RTPSourceClockRate:
@ -106,19 +111,23 @@ struct _RTPSource {
GObject object;
/*< private >*/
RTPSourceCallbacks callbacks;
gpointer user_data;
guint32 ssrc;
gchar *cname;
gint probation;
gboolean validated;
gboolean received_bye;
gchar *bye_reason;
gboolean is_csrc;
gboolean is_sender;
gchar *cname;
gchar *name;
gchar *email;
gchar *phone;
gchar *location;
gchar *tool;
gchar *note;
gboolean received_bye;
gchar *bye_reason;
gboolean have_rtp_from;
GstNetAddress rtp_from;
gboolean have_rtcp_from;
@ -129,6 +138,9 @@ struct _RTPSource {
GQueue *packets;
RTPSourceCallbacks callbacks;
gpointer user_data;
RTPSourceStats stats;
};
@ -147,6 +159,7 @@ void rtp_source_set_as_csrc (RTPSource *src);
void rtp_source_set_rtp_from (RTPSource *src, GstNetAddress *address);
void rtp_source_set_rtcp_from (RTPSource *src, GstNetAddress *address);
/* handling RTP */
GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer);

2
gst/rtpmanager/rtpstats.c
View file

@ -65,7 +65,7 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean sender)
GST_DEBUG ("senders: %f, receivers %f, avg_rtcp %f, sfraction %f",
senders, receivers, avg_rtcp, sfraction);
if (sfraction <= stats->sender_fraction) {
if (senders > 0 && sfraction <= stats->sender_fraction) {
if (sender) {
interval =
(avg_rtcp * senders) / (stats->sender_fraction *

19
gst/rtpmanager/rtpstats.h
View file

@ -92,14 +92,23 @@ typedef struct {
* Stats about a source.
*/
typedef struct {
guint64 packetsreceived;
guint64 prevpacketsreceived;
guint64 octetsreceived;
guint64 bytesreceived;
guint16 max_seqnr;
guint64 packets_received;
guint64 octets_received;
guint64 bytes_received;
guint32 prev_expected;
guint32 prev_received;
guint16 max_seq;
guint32 cycles;
guint32 base_seq;
guint32 bad_seq;
guint32 transit;
guint32 jitter;
guint64 packets_sent;
guint64 octets_sent;
/* when we received stuff */
GstClockTime prev_rtptime;
GstClockTime prev_rtcptime;