rtp: Update codes based on 1.18.4

[platform/upstream/gst-plugins-good.git] / gst / rtp / gstrtpL8pay.c

diff --git a/gst/rtp/gstrtpL8pay.c b/gst/rtp/gstrtpL8pay.c
index 86e7b22..6662cda 100644 (file)
--- a/gst/rtp/gstrtpL8pay.c
+++ b/gst/rtp/gstrtpL8pay.c
@@ -25,13 +25,12 @@
  * Payload raw audio into RTP packets according to RFC 3551.
  * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
  *
- * <refsect2>
- * <title>Example pipeline</title>
+ * ## Example pipeline
+ *
  * |[
  * gst-launch -v audiotestsrc ! audioconvert ! rtpL8pay ! udpsink
  * ]| This example pipeline will payload raw audio. Refer to
  * the rtpL8depay example to depayload and play the RTP stream.
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H
@@ -196,16 +195,10 @@ gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
 
       if (gst_structure_get_int (structure, "channels", &channels)) {
         gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
-      } else {
-        /* Support any number of channels, if not explicitly specified */
-        gst_structure_remove_field (structure, "channels");
       }
 
       if (gst_structure_get_int (structure, "clock-rate", &rate)) {
         gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
-      } else {
-        /* Support any rate, if not explicitly specified */
-        gst_structure_remove_field (structure, "rate");
       }
 
     }