rtp: Update codes based on 1.18.4

[platform/upstream/gst-plugins-good.git] / gst / rtp / gstrtpL8depay.c

diff --git a/gst/rtp/gstrtpL8depay.c b/gst/rtp/gstrtpL8depay.c
index 5b9520a..121a749 100644 (file)
--- a/gst/rtp/gstrtpL8depay.c
+++ b/gst/rtp/gstrtpL8depay.c
@@ -25,13 +25,12 @@
  * Extract raw audio from 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 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L8, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL8depay ! pulsesink
  * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
  * the rtpL8pay example to create the RTP stream.
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H
@@ -187,6 +186,7 @@ gst_rtp_L8_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
             GST_STR_NULL (channel_order), channels));
     /* create default NONE layout */
     gst_rtp_channels_create_default (channels, info->position);
+    info->flags |= GST_AUDIO_FLAG_UNPOSITIONED;
   }
 
   srccaps = gst_audio_info_to_caps (info);