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rtpstats: make bandwidths more configurable

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Add a method to configure the various bandwidths in the session.
This commit is contained in:
Wim Taymans 2010-05-07 16:55:13 +02:00
parent 6eee730c4a
commit 0da5cf2e21
2 changed files with 95 additions and 17 deletions
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99
gst/rtpmanager/rtpstats.c
View file

@ -28,20 +28,85 @@
void
rtp_stats_init_defaults (RTPSessionStats * stats)
{
stats->bandwidth = RTP_STATS_BANDWIDTH;
stats->sender_fraction = RTP_STATS_SENDER_FRACTION;
stats->receiver_fraction = RTP_STATS_RECEIVER_FRACTION;
stats->rtcp_bandwidth = RTP_STATS_RTCP_BANDWIDTH;
rtp_stats_set_bandwidths (stats, -1, -1, -1, -1);
stats->min_interval = RTP_STATS_MIN_INTERVAL;
stats->bye_timeout = RTP_STATS_BYE_TIMEOUT;
}
/**
* rtp_stats_set_bandwidths:
* @stats: an #RTPSessionStats struct
* @rtp_bw: RTP bandwidth
* @rtcp_bw: RTCP bandwidth
* @rs: sender RTCP bandwidth
* @rr: receiver RTCP bandwidth
*
* Configure the bandwidth parameters in the stats. When an input variable is
* set to -1, it will be calculated from the other input variables and from the
* defaults.
*/
void
rtp_stats_set_bandwidths (RTPSessionStats * stats, guint rtp_bw, guint rtcp_bw,
guint rs, guint rr)
{
/* when given, sender and receive bandwidth add up to the total
* rtcp bandwidth */
if (rs != -1 && rr != -1)
rtcp_bw = rs + rr;
/* RTCP is 5% of the RTP bandwidth */
if (rtp_bw == -1 && rtcp_bw != -1)
rtp_bw = rtcp_bw * 20;
else if (rtp_bw != -1 && rtcp_bw == -1)
rtcp_bw = rtp_bw / 20;
else if (rtp_bw == -1 && rtcp_bw == -1) {
/* nothing given, take defaults */
rtp_bw = RTP_STATS_BANDWIDTH;
rtcp_bw = RTP_STATS_RTCP_BANDWIDTH;
}
stats->bandwidth = rtp_bw;
stats->rtcp_bandwidth = rtcp_bw;
/* now figure out the fractions */
if (rs == -1) {
/* rs unknown */
if (rr == -1) {
/* both not given, use defaults */
rs = stats->rtcp_bandwidth * RTP_STATS_SENDER_FRACTION;
rr = stats->rtcp_bandwidth * RTP_STATS_RECEIVER_FRACTION;
} else {
/* rr known, calculate rs */
if (stats->rtcp_bandwidth > rr)
rs = stats->rtcp_bandwidth - rr;
else
rs = 0;
}
} else if (rr == -1) {
/* rs known, calculate rr */
if (stats->rtcp_bandwidth > rs)
rr = stats->rtcp_bandwidth - rs;
else
rr = 0;
}
if (stats->rtcp_bandwidth > 0) {
stats->sender_fraction = ((gdouble) rs) / ((gdouble) stats->rtcp_bandwidth);
stats->receiver_fraction = 1.0 - stats->sender_fraction;
} else {
/* no RTCP bandwidth, set dummy values */
stats->sender_fraction = 0.0;
stats->receiver_fraction = 0.0;
}
GST_DEBUG ("bandwidths: RTP %u, RTCP %u, RS %f, RR %f", stats->bandwidth,
stats->rtcp_bandwidth, stats->sender_fraction, stats->receiver_fraction);
}
/**
* rtp_stats_calculate_rtcp_interval:
* @stats: an #RTPSessionStats struct
* @sender: if we are a sender
* @first: if this is the first time
*
*
* Calculate the RTCP interval. The result of this function is the amount of
* time to wait (in nanoseconds) before sending a new RTCP message.
*
@ -74,16 +139,21 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send,
senders = (gdouble) stats->sender_sources;
rtcp_bw = stats->rtcp_bandwidth;
if (senders <= members * RTP_STATS_SENDER_FRACTION) {
if (senders <= members * stats->sender_fraction) {
if (we_send) {
rtcp_bw *= RTP_STATS_SENDER_FRACTION;
rtcp_bw *= stats->sender_fraction;
n = senders;
} else {
rtcp_bw *= RTP_STATS_RECEIVER_FRACTION;
rtcp_bw *= stats->receiver_fraction;
n -= senders;
}
}
/* no bandwidth for RTCP, return NONE to signal that we don't want to send
* RTCP packets */
if (rtcp_bw <= 0.00001)
return GST_CLOCK_TIME_NONE;
avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0;
/*
* The effective number of sites times the average packet size is
@ -105,7 +175,7 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send,
* rtp_stats_add_rtcp_jitter:
* @stats: an #RTPSessionStats struct
* @interval: an RTCP interval
*
*
* Apply a random jitter to the @interval. @interval is typically obtained with
* rtp_stats_calculate_rtcp_interval().
*
@ -116,7 +186,7 @@ rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, GstClockTime interval)
{
gdouble temp;
/* see RFC 3550 p 30
/* see RFC 3550 p 30
* To compensate for "unconditional reconsideration" converging to a
* value below the intended average.
*/
@ -131,7 +201,7 @@ rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, GstClockTime interval)
/**
* rtp_stats_calculate_bye_interval:
* @stats: an #RTPSessionStats struct
*
*
* Calculate the BYE interval. The result of this function is the amount of
* time to wait (in nanoseconds) before sending a BYE message.
*
@ -156,7 +226,12 @@ rtp_stats_calculate_bye_interval (RTPSessionStats * stats)
* more than that fraction.
*/
members = stats->bye_members;
rtcp_bw = stats->rtcp_bandwidth * RTP_STATS_RECEIVER_FRACTION;
rtcp_bw = stats->rtcp_bandwidth * stats->receiver_fraction;
/* no bandwidth for RTCP, return NONE to signal that we don't want to send
* RTCP packets */
if (rtcp_bw <= 0.0001)
return GST_CLOCK_TIME_NONE;
avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0;
/*

13
gst/rtpmanager/rtpstats.h
View file

@ -127,8 +127,8 @@ typedef struct {
RTPSenderReport sr[2];
} RTPSourceStats;
#define RTP_STATS_BANDWIDTH 64000.0
#define RTP_STATS_RTCP_BANDWIDTH 3000.0
#define RTP_STATS_BANDWIDTH 64000
#define RTP_STATS_RTCP_BANDWIDTH 3200
/*
* Minimum average time between RTCP packets from this site (in
* seconds). This time prevents the reports from `clumping' when
@ -172,10 +172,10 @@ typedef struct {
* Stats kept for a session and used to produce RTCP packet timeouts.
*/
typedef struct {
gdouble bandwidth;
guint bandwidth;
guint rtcp_bandwidth;
gdouble sender_fraction;
gdouble receiver_fraction;
gdouble rtcp_bandwidth;
gdouble min_interval;
GstClockTime bye_timeout;
guint sender_sources;
@ -184,7 +184,10 @@ typedef struct {
guint bye_members;
} RTPSessionStats;
void rtp_stats_init_defaults (RTPSessionStats *stats);
void rtp_stats_init_defaults (RTPSessionStats *stats);
void rtp_stats_set_bandwidths (RTPSessionStats *stats, guint rtp_bw, guint rtcp_bw,
guint rs, guint rr);
GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender, gboolean first);
GstClockTime rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, GstClockTime interval);