rtp: Update codes based on 1.18.4

[platform/upstream/gst-plugins-good.git] / gst / rtp / gstrtpL24depay.c

diff --git a/gst/rtp/gstrtpL24depay.c b/gst/rtp/gstrtpL24depay.c
index 8b28ee8..448dac0 100644 (file)
--- a/gst/rtp/gstrtpL24depay.c
+++ b/gst/rtp/gstrtpL24depay.c
@@ -19,18 +19,18 @@
 
 /**
  * SECTION:element-rtpL24depay
+ * @title: rtpL24depay
  * @see_also: rtpL24pay
  *
  * Extract raw audio from RTP packets according to RFC 3190, section 4.
  * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt
  *
- * <refsect2>
- * <title>Example pipeline</title>
+ * ## Example pipeline
  * |[
  * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
  * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
  * the rtpL24pay example to create the RTP stream.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
@@ -187,6 +187,7 @@ gst_rtp_L24_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);