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session: improve bandwidth recalculation

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Also recalculate bandwidth when one of the source bandwidths changed.
Use the newly calculated bandwidth.
This commit is contained in:
Wim Taymans 2010-09-13 15:29:06 +02:00
parent 6f53a2b240
commit 8381d9788d
1 changed files with 15 additions and 9 deletions
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24
gst/rtpmanager/rtpsession.c
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@ -1385,6 +1385,7 @@ rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
return result;
}
/* should be called with the SESSION lock */
static guint32
rtp_session_create_new_ssrc (RTPSession * sess)
{
@ -1488,6 +1489,7 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
RTPArrivalStats arrival;
guint32 csrcs[16];
guint8 i, count;
guint64 oldrate;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
@ -1512,6 +1514,7 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
prevsender = RTP_SOURCE_IS_SENDER (source);
prevactive = RTP_SOURCE_IS_ACTIVE (source);
oldrate = source->bitrate;
/* copy available csrc for later */
count = gst_rtp_buffer_get_csrc_count (buffer);
@ -1537,6 +1540,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
sess->stats.sender_sources);
}
if (oldrate != source->bitrate)
sess->recalc_bandwidth = TRUE;
if (created)
on_new_ssrc (sess, source);
@ -2022,6 +2027,7 @@ rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
RTPSource *source;
gboolean prevsender;
gboolean valid_packet;
guint64 oldrate;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
@ -2044,12 +2050,15 @@ rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
source->last_rtp_activity = current_time;
prevsender = RTP_SOURCE_IS_SENDER (source);
oldrate = source->bitrate;
/* we use our own source to send */
result = rtp_source_send_rtp (source, data, is_list, running_time);
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
sess->stats.sender_sources++;
if (oldrate != source->bitrate)
sess->recalc_bandwidth = TRUE;
RTP_SESSION_UNLOCK (sess);
return result;
@ -2064,11 +2073,8 @@ invalid_packet:
}
static void
add_bitrates (gpointer key, gpointer value, gpointer user_data)
add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
{
gdouble *bandwidth = user_data;
RTPSource *source = value;
*bandwidth += source->bitrate;
}
@ -2078,8 +2084,8 @@ calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
{
GstClockTime result;
if (sess->recalc_bandwidth || sess->bandwidth == 0) {
/* recalculate bandwidth when it changed */
/* recalculate bandwidth when it changed */
if (sess->recalc_bandwidth) {
gdouble bandwidth;
if (sess->bandwidth > 0)
@ -2088,15 +2094,15 @@ calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
/* If it is <= 0, then try to estimate the actual bandwidth */
bandwidth = sess->source->bitrate;
g_hash_table_foreach (sess->cnames, add_bitrates, &bandwidth);
g_hash_table_foreach (sess->cnames, (GHFunc) add_bitrates, &bandwidth);
bandwidth /= 8.0;
}
if (bandwidth == 0)
bandwidth = RTP_STATS_BANDWIDTH;
rtp_stats_set_bandwidths (&sess->stats, sess->bandwidth,
rtp_stats_set_bandwidths (&sess->stats, bandwidth,
sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
sess->recalc_bandwidth = FALSE;
}