rtp: Update codes based on 1.18.4

[platform/upstream/gst-plugins-good.git] / gst / rtp / gstrtpL16pay.c

diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index af98ba9..7e358d3 100644 (file)
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -13,8 +13,24 @@
  *
  * You should have received a copy of the GNU Library General Public
  * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-rtpL16pay
+ * @title: rtpL16pay
+ * @see_also: rtpL16depay
+ *
+ * Payload raw audio into RTP packets according to RFC 3551.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
+ *
+ * ## Example pipeline
+ * |[
+ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
+ * ]| This example pipeline will payload raw audio. Refer to
+ * the rtpL16depay example to depayload and play the RTP stream.
+ *
  */
 
 #ifdef HAVE_CONFIG_H
@@ -37,7 +53,7 @@ GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) S16_BE, "
+        "format = (string) S16BE, "
         "layout = (string) interleaved, "
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
     );
@@ -88,13 +104,13 @@ gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
   gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps;
   gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
 
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&gst_rtp_L16_pay_src_template));
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&gst_rtp_L16_pay_sink_template));
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_L16_pay_src_template);
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_L16_pay_sink_template);
 
-  gst_element_class_set_details_simple (gstelement_class, "RTP audio payloader",
-      "Codec/Payloader/Network/RTP",
+  gst_element_class_set_static_metadata (gstelement_class,
+      "RTP audio payloader", "Codec/Payloader/Network/RTP",
       "Payload-encode Raw audio into RTP packets (RFC 3551)",
       "Wim Taymans <wim.taymans@gmail.com>");
 
@@ -176,9 +192,9 @@ gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
   GstCaps *otherpadcaps;
   GstCaps *caps;
 
-  otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
-  caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+  caps = gst_pad_get_pad_template_caps (pad);
 
+  otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
   if (otherpadcaps) {
     if (!gst_caps_is_empty (otherpadcaps)) {
       GstStructure *structure;
@@ -187,20 +203,21 @@ gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
       gint rate;
 
       structure = gst_caps_get_structure (otherpadcaps, 0);
+      caps = gst_caps_make_writable (caps);
 
       if (gst_structure_get_int (structure, "channels", &channels)) {
         gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
       } else if (gst_structure_get_int (structure, "payload", &pt)) {
-        if (pt == 10)
+        if (pt == GST_RTP_PAYLOAD_L16_STEREO)
           gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
-        else if (pt == 11)
+        else if (pt == GST_RTP_PAYLOAD_L16_MONO)
           gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
       }
 
       if (gst_structure_get_int (structure, "clock-rate", &rate)) {
         gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
       } else if (gst_structure_get_int (structure, "payload", &pt)) {
-        if (pt == 10 || pt == 11)
+        if (pt == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO)
           gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
       }