[gst-cvs] gst-plugins-good: rtph264pay: scale spspps_interval to milliseconds

[Wim Taymans]

Module: gst-plugins-good
Branch: master
Commit: 1f6b06ce6674a3a91db7d6b443e2223f93ef3872
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1f6b06ce6674a3a91db7d6b443e2223f93ef3872

Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date:   Wed Jan 20 16:28:31 2010 +0100

rtph264pay: scale spspps_interval to milliseconds

The spspps_interval is kept in seconds. Convert it to milliseconds before
comparing it to another value in milliseconds.

---

 gst/rtp/gstrtph264pay.c |    5 +++--
 1 files changed, 3 insertions(+), 2 deletions(-)

diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
index 701d6fa..fbb7132 100644
--- a/gst/rtp/gstrtph264pay.c
+++ b/gst/rtp/gstrtph264pay.c
@@ -655,7 +655,7 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload, guint8 * data,
       GST_DEBUG_OBJECT (rtph264pay, "interval since last SPS/PPS %ums", diff);
 
       /* bigger than interval, queue SPS/PPS */
-      if (diff >= rtph264pay->spspps_interval) {
+      if (diff >= (rtph264pay->spspps_interval * GST_MSECOND)) {
         GST_DEBUG_OBJECT (rtph264pay, "time to send SPS/PPS");
         send_spspps = TRUE;
       }
@@ -667,7 +667,8 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload, guint8 * data,
   }
 
   if (send_spspps) {
-    /* we need to send SPS/PPS now first */
+    /* we need to send SPS/PPS now first. FIXME, don't use the timestamp for
+     * checking when we need to send SPS/PPS but convert to running_time first. */
     ret = gst_rtp_h264_pay_send_sps_pps (basepayload, rtph264pay, timestamp);
     if (ret != GST_FLOW_OK)
       return ret;