mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-20 08:41:07 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtpsource: refactor bitrate estimation

Browse source 
Don't reuse the same variable we need for stats for the bitrate estimation
because we're updating it.
Refactor the bitrate estimation code so that both sender and receivers use the
same code path.
This commit is contained in:
Wim Taymans 2010-03-02 12:39:20 +01:00
parent a0a6d4ff3b
commit c971d1a9ab
2 changed files with 39 additions and 55 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

93
gst/rtpmanager/rtpsource.c
View file

@ -886,6 +886,40 @@ init_seq (RTPSource * src, guint16 seq)
GST_DEBUG ("base_seq %d", seq); GST_DEBUG ("base_seq %d", seq);
} }
static void
do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
guint64 * bytes_handled)
{
guint64 elapsed;
if (src->prev_rtime) {
elapsed = running_time - src->prev_rtime;
if (elapsed > (G_GINT64_CONSTANT (1) << 31)) {
guint64 rate;
rate =
gst_util_uint64_scale (*bytes_handled, elapsed,
(G_GINT64_CONSTANT (1) << 29));
GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
if (src->bitrate == 0)
src->bitrate = rate;
else
src->bitrate = ((src->bitrate * 3) + rate) / 4;
src->prev_rtime = running_time;
*bytes_handled = 0;
}
} else {
GST_LOG ("Reset bitrate measurement");
src->prev_rtime = running_time;
src->bitrate = 0;
}
}
/** /**
* rtp_source_process_rtp: * rtp_source_process_rtp:
* @src: an #RTPSource * @src: an #RTPSource
@ -903,7 +937,6 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
guint16 seqnr, udelta; guint16 seqnr, udelta;
RTPSourceStats *stats; RTPSourceStats *stats;
guint16 expected; guint16 expected;
guint64 elapsed;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR); g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR); g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
@ -981,37 +1014,13 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
src->stats.octets_received += arrival->payload_len; src->stats.octets_received += arrival->payload_len;
src->stats.bytes_received += arrival->bytes; src->stats.bytes_received += arrival->bytes;
src->stats.packets_received++; src->stats.packets_received++;
/* for the bitrate estimation */
src->bytes_received += arrival->bytes;
/* the source that sent the packet must be a sender */ /* the source that sent the packet must be a sender */
src->is_sender = TRUE; src->is_sender = TRUE;
src->validated = TRUE; src->validated = TRUE;
if (src->prev_rtime) { do_bitrate_estimation (src, arrival->running_time, &src->bytes_received);
elapsed = arrival->running_time - src->prev_rtime;
if (elapsed > (G_GINT64_CONSTANT (1) << 31)) {
guint64 rate;
rate =
gst_util_uint64_scale (src->stats.bytes_received, elapsed,
(G_GINT64_CONSTANT (1) << 29));
GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
", rate %" G_GUINT64_FORMAT, elapsed, src->stats.bytes_received,
rate);
if (src->bitrate == 0)
src->bitrate = rate;
else
src->bitrate = ((src->bitrate * 3) + rate) / 4;
src->prev_rtime = arrival->running_time;
src->stats.bytes_received = 0;
}
} else {
GST_LOG ("Reset bitrate measurement");
src->prev_rtime = arrival->running_time;
src->bitrate = 0;
}
GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT, GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
seqnr, src->stats.packets_received, src->stats.octets_received); seqnr, src->stats.packets_received, src->stats.octets_received);
@ -1094,7 +1103,6 @@ rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
guint32 rtptime; guint32 rtptime;
guint64 ext_rtptime; guint64 ext_rtptime;
guint64 rt_diff, rtp_diff; guint64 rt_diff, rtp_diff;
guint64 elapsed;
GstBufferList *list = NULL; GstBufferList *list = NULL;
GstBuffer *buffer = NULL; GstBuffer *buffer = NULL;
guint packets; guint packets;
@ -1133,32 +1141,7 @@ rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
src->stats.octets_sent += len; src->stats.octets_sent += len;
src->bytes_sent += len; src->bytes_sent += len;
if (src->prev_rtime) { do_bitrate_estimation (src, running_time, &src->bytes_sent);
elapsed = running_time - src->prev_rtime;
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_rtime = running_time;
src->bytes_sent = 0;
}
} else {
GST_LOG ("Reset bitrate measurement");
src->prev_rtime = running_time;
src->bitrate = 0;
}
if (is_list) { if (is_list) {
rtptime = gst_rtp_buffer_list_get_timestamp (list); rtptime = gst_rtp_buffer_list_get_timestamp (list);

1
gst/rtpmanager/rtpsource.h
View file

@ -143,6 +143,7 @@ struct _RTPSource {
guint64 bitrate; guint64 bitrate;
GstClockTime prev_rtime; GstClockTime prev_rtime;
guint64 bytes_sent; guint64 bytes_sent;
guint64 bytes_received;
GQueue *packets; GQueue *packets;