From: Wim Taymans <wim.taymans@collabora.co.uk>
Date: Thu, 10 Nov 2011 16:23:47 +0000 (+0100)
Subject: update for changed base classes
X-Git-Tag: 1.19.3~509^2~7550
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=fbaf216d2554897b09f00420cb8cb6a107ffc2b3;p=platform%2Fupstream%2Fgstreamer.git

update for changed base classes
---

diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index e13e135..7259931 100644
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -140,7 +140,7 @@ gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
   else
     order = NULL;
 
-  gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate);
+  gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "L16", rate);
   params = g_strdup_printf ("%d", channels);
 
   if (!order && channels > 2) {
@@ -149,11 +149,11 @@ gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
   }
 
   if (order && order->name) {
-    res = gst_basertppayload_set_outcaps (basepayload,
+    res = gst_base_rtp_payload_set_outcaps (basepayload,
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
         channels, "channel-order", G_TYPE_STRING, order->name, NULL);
   } else {
-    res = gst_basertppayload_set_outcaps (basepayload,
+    res = gst_base_rtp_payload_set_outcaps (basepayload,
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
         channels, NULL);
   }
diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c
index 851b92d..d561455 100644
--- a/gst/rtp/gstrtpac3pay.c
+++ b/gst/rtp/gstrtpac3pay.c
@@ -136,8 +136,8 @@ gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   if (!gst_structure_get_int (structure, "rate", &rate))
     rate = 90000;               /* default */
 
-  gst_basertppayload_set_options (payload, "audio", TRUE, "AC3", rate);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "audio", TRUE, "AC3", rate);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
@@ -310,7 +310,7 @@ gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay)
     GST_BUFFER_TIMESTAMP (outbuf) = rtpac3pay->first_ts;
     GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
 
-    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf);
+    ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf);
   }
 
   return ret;
@@ -385,7 +385,7 @@ gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   /* if this buffer is going to overflow the packet, flush what we
    * have. */
-  if (gst_basertppayload_is_filled (basepayload,
+  if (gst_base_rtp_payload_is_filled (basepayload,
           packet_len, rtpac3pay->duration + duration)) {
     ret = gst_rtp_ac3_pay_flush (rtpac3pay);
     avail = 0;
diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c
index 5454b69..930779c 100644
--- a/gst/rtp/gstrtpamrpay.c
+++ b/gst/rtp/gstrtpamrpay.c
@@ -160,12 +160,12 @@ gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
     goto wrong_type;
 
   if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
-    gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
+    gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
   else
-    gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR-WB",
+    gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "AMR-WB",
         16000);
 
-  res = gst_basertppayload_set_outcaps (basepayload,
+  res = gst_base_rtp_payload_set_outcaps (basepayload,
       "encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
       /* don't set the defaults
        *
@@ -376,7 +376,7 @@ gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   gst_rtp_buffer_unmap (&rtp);
 
-  ret = gst_basertppayload_push (basepayload, outbuf);
+  ret = gst_base_rtp_payload_push (basepayload, outbuf);
 
   return ret;
 
diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c
index 2066d9c..6388653 100644
--- a/gst/rtp/gstrtpbvpay.c
+++ b/gst/rtp/gstrtpbvpay.c
@@ -125,11 +125,11 @@ gst_rtp_bv_pay_sink_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
     goto wrong_mode;
 
   if (mode == 16) {
-    gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV16",
+    gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "BV16",
         8000);
     basertppayload->clock_rate = 8000;
   } else {
-    gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV32",
+    gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "BV32",
         16000);
     basertppayload->clock_rate = 16000;
   }
diff --git a/gst/rtp/gstrtpceltpay.c b/gst/rtp/gstrtpceltpay.c
index 176de7e..096b1a9 100644
--- a/gst/rtp/gstrtpceltpay.c
+++ b/gst/rtp/gstrtpceltpay.c
@@ -251,10 +251,10 @@ gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay,
 
   payload = GST_BASE_RTP_PAYLOAD (rtpceltpay);
 
-  gst_basertppayload_set_options (payload, "audio", FALSE, "CELT", rate);
+  gst_base_rtp_payload_set_options (payload, "audio", FALSE, "CELT", rate);
   cstr = g_strdup_printf ("%d", nb_channels);
   fsstr = g_strdup_printf ("%d", frame_size);
-  res = gst_basertppayload_set_outcaps (payload, "encoding-params",
+  res = gst_base_rtp_payload_set_outcaps (payload, "encoding-params",
       G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL);
   g_free (cstr);
   g_free (fsstr);
@@ -354,7 +354,7 @@ gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
   rtpceltpay->sbytes = 0;
   rtpceltpay->qduration = 0;
 
-  ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpceltpay), outbuf);
+  ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpceltpay), outbuf);
 
   return ret;
 }
@@ -415,7 +415,7 @@ gst_rtp_celt_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
 
-  if (gst_basertppayload_is_filled (basepayload, packet_len, packet_dur)) {
+  if (gst_base_rtp_payload_is_filled (basepayload, packet_len, packet_dur)) {
     /* size or duration would overflow the packet, flush the queued data */
     ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);
   }
diff --git a/gst/rtp/gstrtpdvpay.c b/gst/rtp/gstrtpdvpay.c
index 10690ab..0d0cc18 100644
--- a/gst/rtp/gstrtpdvpay.c
+++ b/gst/rtp/gstrtpdvpay.c
@@ -221,15 +221,15 @@ gst_dv_pay_negotiate (GstRTPDVPay * rtpdvpay, guint8 * data, gsize size)
     default:
       break;
   }
-  gst_basertppayload_set_options (GST_BASE_RTP_PAYLOAD (rtpdvpay), media, TRUE,
-      "DV", 90000);
+  gst_base_rtp_payload_set_options (GST_BASE_RTP_PAYLOAD (rtpdvpay), media,
+      TRUE, "DV", 90000);
 
   if (audio_bundled) {
-    res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
+    res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
         "encode", G_TYPE_STRING, encode,
         "audio", G_TYPE_STRING, "bundled", NULL);
   } else {
-    res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
+    res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpdvpay),
         "encode", G_TYPE_STRING, encode, NULL);
   }
   return res;
@@ -360,7 +360,7 @@ gst_rtp_dv_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
       /* Push out the created piece, and check for errors. */
       gst_rtp_buffer_unmap (&rtp);
-      ret = gst_basertppayload_push (basepayload, outbuf);
+      ret = gst_base_rtp_payload_push (basepayload, outbuf);
       if (ret != GST_FLOW_OK)
         break;
 
diff --git a/gst/rtp/gstrtpg722pay.c b/gst/rtp/gstrtpg722pay.c
index d07abfe..bc19a5d 100644
--- a/gst/rtp/gstrtpg722pay.c
+++ b/gst/rtp/gstrtpg722pay.c
@@ -132,7 +132,7 @@ gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
    * RFC 3551 although the sampling rate is 16000 Hz */
   clock_rate = 8000;
 
-  gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
+  gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "G722",
       clock_rate);
   params = g_strdup_printf ("%d", channels);
 
@@ -142,11 +142,11 @@ gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
   }
 
   if (order && order->name) {
-    res = gst_basertppayload_set_outcaps (basepayload,
+    res = gst_base_rtp_payload_set_outcaps (basepayload,
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
         channels, "channel-order", G_TYPE_STRING, order->name, NULL);
   } else {
-    res = gst_basertppayload_set_outcaps (basepayload,
+    res = gst_base_rtp_payload_set_outcaps (basepayload,
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
         channels, NULL);
   }
diff --git a/gst/rtp/gstrtpg723pay.c b/gst/rtp/gstrtpg723pay.c
index 2621c55..e870ab5 100644
--- a/gst/rtp/gstrtpg723pay.c
+++ b/gst/rtp/gstrtpg723pay.c
@@ -107,7 +107,7 @@ gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
   pay->adapter = gst_adapter_new ();
 
   payload->pt = GST_RTP_PAYLOAD_G723;
-  gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
+  gst_base_rtp_payload_set_options (payload, "audio", FALSE, "G723", 8000);
 }
 
 static void
@@ -138,7 +138,7 @@ gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
   payload->pt = pt;
   payload->dynamic = pt != GST_RTP_PAYLOAD_G723;
 
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
@@ -178,7 +178,7 @@ gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
   }
   gst_rtp_buffer_unmap (&rtp);
 
-  ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
+  ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
 
   return ret;
 }
@@ -229,7 +229,7 @@ gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
   packet_dur = pay->duration + G723_FRAME_DURATION;
   packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
 
-  if (gst_basertppayload_is_filled (payload, packet_len, packet_dur)) {
+  if (gst_base_rtp_payload_is_filled (payload, packet_len, packet_dur)) {
     /* size or duration would overflow the packet, flush the queued data */
     ret = gst_rtp_g723_pay_flush (pay);
   }
diff --git a/gst/rtp/gstrtpg726pay.c b/gst/rtp/gstrtpg726pay.c
index 12b772f..db76d03 100644
--- a/gst/rtp/gstrtpg726pay.c
+++ b/gst/rtp/gstrtpg726pay.c
@@ -237,8 +237,9 @@ gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
     GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2);
   }
 
-  gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "audio", TRUE, encoding_name,
+      8000);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   g_free (encoding_name);
 
diff --git a/gst/rtp/gstrtpg729pay.c b/gst/rtp/gstrtpg729pay.c
index 746b2f9..c827c31 100644
--- a/gst/rtp/gstrtpg729pay.c
+++ b/gst/rtp/gstrtpg729pay.c
@@ -121,7 +121,7 @@ gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
   GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
 
   payload->pt = GST_RTP_PAYLOAD_G729;
-  gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000);
+  gst_base_rtp_payload_set_options (payload, "audio", FALSE, "G729", 8000);
 
   pay->adapter = gst_adapter_new ();
 }
@@ -150,7 +150,7 @@ gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
   payload->pt = pt;
   payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
 
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
@@ -199,7 +199,7 @@ gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay,
   }
   gst_rtp_buffer_unmap (&rtp);
 
-  ret = gst_basertppayload_push (basepayload, outbuf);
+  ret = gst_base_rtp_payload_push (basepayload, outbuf);
 
   return ret;
 }
diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c
index 5a8061f..1fbe424 100644
--- a/gst/rtp/gstrtpgsmpay.c
+++ b/gst/rtp/gstrtpgsmpay.c
@@ -107,8 +107,8 @@ gst_rtp_gsm_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   if (strcmp ("audio/x-gsm", stname))
     goto invalid_type;
 
-  gst_basertppayload_set_options (payload, "audio", FALSE, "GSM", 8000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "audio", FALSE, "GSM", 8000);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 
@@ -168,7 +168,7 @@ gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * basepayload,
   GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %d",
       gst_buffer_get_size (outbuf));
 
-  ret = gst_basertppayload_push (basepayload, outbuf);
+  ret = gst_base_rtp_payload_push (basepayload, outbuf);
 
   return ret;
 
diff --git a/gst/rtp/gstrtpgstpay.c b/gst/rtp/gstrtpgstpay.c
index 571a7a8..3d405d9 100644
--- a/gst/rtp/gstrtpgstpay.c
+++ b/gst/rtp/gstrtpgstpay.c
@@ -111,9 +111,10 @@ gst_rtp_gst_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   capsenc = g_base64_encode ((guchar *) capsstr, strlen (capsstr));
   g_free (capsstr);
 
-  gst_basertppayload_set_options (payload, "application", TRUE, "X-GST", 90000);
+  gst_base_rtp_payload_set_options (payload, "application", TRUE, "X-GST",
+      90000);
   res =
-      gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsenc,
+      gst_base_rtp_payload_set_outcaps (payload, "caps", G_TYPE_STRING, capsenc,
       NULL);
   g_free (capsenc);
 
@@ -203,7 +204,7 @@ gst_rtp_gst_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
     GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
 
-    ret = gst_basertppayload_push (basepayload, outbuf);
+    ret = gst_base_rtp_payload_push (basepayload, outbuf);
   }
   gst_buffer_unmap (buffer, data, size);
   gst_buffer_unref (buffer);
diff --git a/gst/rtp/gstrtph263depay.c b/gst/rtp/gstrtph263depay.c
index 13a2cca..239a654 100644
--- a/gst/rtp/gstrtph263depay.c
+++ b/gst/rtp/gstrtph263depay.c
@@ -253,7 +253,7 @@ gst_rtp_h263_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
     if (!F && payload_len > 4 && (GST_READ_UINT32_BE (payload) >> 10 == 0x20)) {
       GST_DEBUG ("Mode A with PSC => frame start");
       rtph263depay->start = TRUE;
-      if (!!(payload[4] & 0x02) != I) {
+      if (! !(payload[4] & 0x02) != I) {
         GST_DEBUG ("Wrong Picture Coding Type Flag in rtp header");
         I = !I;
       }
@@ -307,7 +307,6 @@ skip:
     if (rtph263depay->start) {
       /* frame is completed */
       guint avail;
-      guint32 timestamp;
 
       if (rtph263depay->offset) {
         /* push in the leftover */
@@ -326,8 +325,7 @@ skip:
 
       GST_DEBUG ("Pushing out a buffer of %d bytes", avail);
 
-      timestamp = gst_rtp_buffer_get_timestamp (&rtp);
-      gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
+      gst_base_rtp_depayload_push (depayload, outbuf);
       rtph263depay->offset = 0;
       rtph263depay->leftover = 0;
       rtph263depay->start = FALSE;
diff --git a/gst/rtp/gstrtph263pay.c b/gst/rtp/gstrtph263pay.c
index 8bb5efe..1efbb2d 100644
--- a/gst/rtp/gstrtph263pay.c
+++ b/gst/rtp/gstrtph263pay.c
@@ -464,8 +464,8 @@ gst_rtp_h263_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   gboolean res;
 
   payload->pt = GST_RTP_PAYLOAD_H263;
-  gst_basertppayload_set_options (payload, "video", TRUE, "H263", 90000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "video", TRUE, "H263", 90000);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
@@ -1279,7 +1279,7 @@ gst_rtp_h263_pay_push (GstRtpH263Pay * rtph263pay,
   gst_rtp_buffer_unmap (&rtp);
 
   ret =
-      gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263pay),
+      gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtph263pay),
       package->outbuf);
   GST_DEBUG ("Package pushed, returning");
 
diff --git a/gst/rtp/gstrtph263ppay.c b/gst/rtp/gstrtph263ppay.c
index 73f1a33..4ddfbb6 100644
--- a/gst/rtp/gstrtph263ppay.c
+++ b/gst/rtp/gstrtph263ppay.c
@@ -198,9 +198,9 @@ gst_rtp_h263p_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   if (!encoding_name)
     encoding_name = g_strdup ("H263-1998");
 
-  gst_basertppayload_set_options (payload, "video", TRUE,
+  gst_base_rtp_payload_set_options (payload, "video", TRUE,
       (gchar *) encoding_name, 90000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
   g_free (encoding_name);
 
   return res;
@@ -719,7 +719,8 @@ gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay)
 
     gst_adapter_flush (rtph263ppay->adapter, towrite);
 
-    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
+    ret =
+        gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
 
     avail -= towrite;
     fragmented = TRUE;
diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
index 8ee8bf8..d300e9a 100644
--- a/gst/rtp/gstrtph264pay.c
+++ b/gst/rtp/gstrtph264pay.c
@@ -118,7 +118,7 @@ static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
     GstBuffer * buffer);
 static gboolean gst_rtp_h264_pay_handle_event (GstBaseRTPPayload * payload,
     GstEvent * event);
-static GstStateChangeReturn gst_basertppayload_change_state (GstElement *
+static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement *
     element, GstStateChange transition);
 
 #define gst_rtp_h264_pay_parent_class parent_class
@@ -188,7 +188,7 @@ gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass)
       "Laurent Glayal <spglegle@yahoo.fr>");
 
   gstelement_class->change_state =
-      GST_DEBUG_FUNCPTR (gst_basertppayload_change_state);
+      GST_DEBUG_FUNCPTR (gst_rtp_h264_pay_change_state);
 
   gstbasertppayload_class->get_caps = gst_rtp_h264_pay_getcaps;
   gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
@@ -406,7 +406,7 @@ gst_rtp_h264_pay_set_sps_pps (GstBaseRTPPayload * basepayload)
   /* profile is 24 bit. Force it to respect the limit */
   profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
   /* combine into output caps */
-  res = gst_basertppayload_set_outcaps (basepayload,
+  res = gst_base_rtp_payload_set_outcaps (basepayload,
       "sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
   g_string_free (sprops, TRUE);
   g_free (profile);
@@ -431,7 +431,7 @@ gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
 
   /* we can only set the output caps when we found the sprops and profile
    * NALs */
-  gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
+  gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "H264", 90000);
 
   alignment = gst_structure_get_string (str, "alignment");
   if (alignment && !strcmp (alignment, "au"))
@@ -944,7 +944,7 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
       gst_buffer_list_add (list, paybuf);
 
       /* push the list to the next element in the pipe */
-      ret = gst_basertppayload_push_list (basepayload, list);
+      ret = gst_base_rtp_payload_push_list (basepayload, list);
     } else
 #endif
     {
@@ -953,7 +953,7 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
       memcpy (payload, data, size);
       gst_rtp_buffer_unmap (&rtp);
 
-      ret = gst_basertppayload_push (basepayload, outbuf);
+      ret = gst_base_rtp_payload_push (basepayload, outbuf);
     }
   } else {
     /* fragmentation Units FU-A */
@@ -1052,7 +1052,7 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
             "recorded %d payload bytes into packet iteration=%d",
             limitedSize + 2, ii);
 
-        ret = gst_basertppayload_push (basepayload, outbuf);
+        ret = gst_base_rtp_payload_push (basepayload, outbuf);
         if (ret != GST_FLOW_OK)
           break;
       }
@@ -1067,7 +1067,7 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
     if (rtph264pay->buffer_list) {
       /* free iterator and push the whole buffer list at once */
       gst_buffer_list_iterator_free (it);
-      ret = gst_basertppayload_push_list (basepayload, list);
+      ret = gst_base_rtp_payload_push_list (basepayload, list);
     }
 #endif
   }
@@ -1218,7 +1218,7 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
       if (rtph264pay->sprop_parameter_sets != NULL) {
         /* explicitly set profile and sprop, use those */
         if (rtph264pay->update_caps) {
-          if (!gst_basertppayload_set_outcaps (basepayload,
+          if (!gst_base_rtp_payload_set_outcaps (basepayload,
                   "sprop-parameter-sets", G_TYPE_STRING,
                   rtph264pay->sprop_parameter_sets, NULL))
             goto caps_rejected;
@@ -1352,8 +1352,7 @@ gst_rtp_h264_pay_handle_event (GstBaseRTPPayload * payload, GstEvent * event)
 }
 
 static GstStateChangeReturn
-gst_basertppayload_change_state (GstElement * element,
-    GstStateChange transition)
+gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
 {
   GstStateChangeReturn ret;
   GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (element);
diff --git a/gst/rtp/gstrtpilbcpay.c b/gst/rtp/gstrtpilbcpay.c
index 3460e7f..5914f99 100644
--- a/gst/rtp/gstrtpilbcpay.c
+++ b/gst/rtp/gstrtpilbcpay.c
@@ -129,14 +129,15 @@ gst_rtp_ilbc_pay_sink_setcaps (GstBaseRTPPayload * basertppayload,
   if (mode != 20 && mode != 30)
     goto wrong_mode;
 
-  gst_basertppayload_set_options (basertppayload, "audio", TRUE, "ILBC", 8000);
+  gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "ILBC",
+      8000);
   /* set options for this frame based audio codec */
   gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload,
       mode, mode == 30 ? 50 : 38);
 
   mode_str = g_strdup_printf ("%d", mode);
   ret =
-      gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_STRING,
+      gst_base_rtp_payload_set_outcaps (basertppayload, "mode", G_TYPE_STRING,
       mode_str, NULL);
   g_free (mode_str);
 
diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c
index d615ea9..48a58c8 100644
--- a/gst/rtp/gstrtpj2kpay.c
+++ b/gst/rtp/gstrtpj2kpay.c
@@ -174,9 +174,9 @@ gst_rtp_j2k_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
     pay->width = width;
   }
 
-  gst_basertppayload_set_options (basepayload, "video", TRUE, "JPEG2000",
+  gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "JPEG2000",
       90000);
-  res = gst_basertppayload_set_outcaps (basepayload, NULL);
+  res = gst_base_rtp_payload_set_outcaps (basepayload, NULL);
 
   return res;
 }
@@ -520,7 +520,7 @@ gst_rtp_j2k_pay_handle_buffer (GstBaseRTPPayload * basepayload,
         memcpy (header + HEADER_SIZE, &data[offset], data_size);
         gst_rtp_buffer_unmap (&rtp);
 
-        ret = gst_basertppayload_push (basepayload, outbuf);
+        ret = gst_base_rtp_payload_push (basepayload, outbuf);
         if (ret != GST_FLOW_OK)
           goto done;
       }
@@ -542,7 +542,7 @@ done:
   if (pay->buffer_list) {
     /* free iterator and push the whole buffer list at once */
     gst_buffer_list_iterator_free (it);
-    ret = gst_basertppayload_push_list (basepayload, list);
+    ret = gst_base_rtp_payload_push_list (basepayload, list);
   }
 #endif
 
diff --git a/gst/rtp/gstrtpjpegpay.c b/gst/rtp/gstrtpjpegpay.c
index 3c8ca57..4a8a202 100644
--- a/gst/rtp/gstrtpjpegpay.c
+++ b/gst/rtp/gstrtpjpegpay.c
@@ -318,8 +318,8 @@ gst_rtp_jpeg_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
   }
   pay->width = width / 8;
 
-  gst_basertppayload_set_options (basepayload, "video", TRUE, "JPEG", 90000);
-  res = gst_basertppayload_set_outcaps (basepayload, NULL);
+  gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "JPEG", 90000);
+  res = gst_base_rtp_payload_set_outcaps (basepayload, NULL);
 
   return res;
 
@@ -819,7 +819,7 @@ gst_rtp_jpeg_pay_handle_buffer (GstBaseRTPPayload * basepayload,
       /* and add to list */
       gst_buffer_list_insert (list, -1, outbuf);
     } else {
-      ret = gst_basertppayload_push (basepayload, outbuf);
+      ret = gst_base_rtp_payload_push (basepayload, outbuf);
       if (ret != GST_FLOW_OK)
         break;
     }
@@ -832,7 +832,7 @@ gst_rtp_jpeg_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   if (pay->buffer_list) {
     /* push the whole buffer list at once */
-    ret = gst_basertppayload_push_list (basepayload, list);
+    ret = gst_base_rtp_payload_push_list (basepayload, list);
   }
 
   gst_buffer_unmap (buffer, bdata, -1);
diff --git a/gst/rtp/gstrtpmp2tpay.c b/gst/rtp/gstrtpmp2tpay.c
index 8fc2c89..f38394b 100644
--- a/gst/rtp/gstrtpmp2tpay.c
+++ b/gst/rtp/gstrtpmp2tpay.c
@@ -108,8 +108,8 @@ gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 {
   gboolean res;
 
-  gst_basertppayload_set_options (payload, "video", TRUE, "MP2T", 90000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "video", TRUE, "MP2T", 90000);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
@@ -140,7 +140,7 @@ gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
   GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d",
       gst_buffer_get_size (outbuf));
 
-  ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
+  ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
 
   /* flush the adapter content */
   gst_adapter_flush (rtpmp2tpay->adapter, avail);
@@ -178,7 +178,7 @@ gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   /* if this buffer is going to overflow the packet, flush what we
    * have. */
-  if (gst_basertppayload_is_filled (basepayload,
+  if (gst_base_rtp_payload_is_filled (basepayload,
           packet_len, rtpmp2tpay->duration + duration)) {
     ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
     rtpmp2tpay->first_ts = timestamp;
diff --git a/gst/rtp/gstrtpmp4apay.c b/gst/rtp/gstrtpmp4apay.c
index 59c7e60..d7a42c1 100644
--- a/gst/rtp/gstrtpmp4apay.c
+++ b/gst/rtp/gstrtpmp4apay.c
@@ -228,7 +228,7 @@ gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
   gst_value_set_buffer (&v, rtpmp4apay->config);
   config = gst_value_serialize (&v);
 
-  res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
+  res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
       "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
 
   g_value_unset (&v);
@@ -321,7 +321,7 @@ gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
     GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
   }
 
-  gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM",
+  gst_base_rtp_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
       rtpmp4apay->rate);
 
   res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
@@ -423,7 +423,7 @@ gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload * basepayload,
     /* copy incomming timestamp (if any) to outgoing buffers */
     GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
 
-    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf);
+    ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf);
 
     fragmented = TRUE;
   }
diff --git a/gst/rtp/gstrtpmp4gpay.c b/gst/rtp/gstrtpmp4gpay.c
index 1514911..5b81c1c 100644
--- a/gst/rtp/gstrtpmp4gpay.c
+++ b/gst/rtp/gstrtpmp4gpay.c
@@ -370,10 +370,10 @@ gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
 
   /* hmm, silly */
   if (rtpmp4gpay->params) {
-    res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+    res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
         "encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
   } else {
-    res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+    res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
         MP4GCAPS);
   }
 
@@ -432,7 +432,7 @@ gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   if (media_type == NULL)
     goto config_failed;
 
-  gst_basertppayload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
+  gst_base_rtp_payload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
       rtpmp4gpay->rate);
 
   res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
@@ -545,7 +545,7 @@ gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
       rtpmp4gpay->offset += rtpmp4gpay->frame_len;
     }
 
-    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
+    ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
 
     avail -= payload_len;
   }
diff --git a/gst/rtp/gstrtpmp4vpay.c b/gst/rtp/gstrtpmp4vpay.c
index 21b8d0d..be64144 100644
--- a/gst/rtp/gstrtpmp4vpay.c
+++ b/gst/rtp/gstrtpmp4vpay.c
@@ -180,7 +180,7 @@ gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
   gst_value_set_buffer (&v, rtpmp4vpay->config);
   config = gst_value_serialize (&v);
 
-  res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
+  res = gst_base_rtp_payload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
       "profile-level-id", G_TYPE_STRING, profile,
       "config", G_TYPE_STRING, config, NULL);
 
@@ -202,7 +202,7 @@ gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 
   rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
 
-  gst_basertppayload_set_options (payload, "video", TRUE, "MP4V-ES",
+  gst_base_rtp_payload_set_options (payload, "video", TRUE, "MP4V-ES",
       rtpmp4vpay->rate);
 
   res = TRUE;
@@ -310,14 +310,16 @@ gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
       /* add to list */
       gst_buffer_list_insert (list, -1, outbuf);
     } else {
-      ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
+      ret =
+          gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
     }
   }
 
   if (rtpmp4vpay->buffer_list) {
     /* push the whole buffer list at once */
     ret =
-        gst_basertppayload_push_list (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), list);
+        gst_base_rtp_payload_push_list (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
+        list);
   }
 
   return ret;
@@ -561,7 +563,7 @@ gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * basepayload,
   /* get packet length of data and see if we exceeded MTU. */
   packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
 
-  if (gst_basertppayload_is_filled (basepayload,
+  if (gst_base_rtp_payload_is_filled (basepayload,
           packet_len, rtpmp4vpay->duration + duration)) {
     ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
     rtpmp4vpay->first_timestamp = timestamp;
diff --git a/gst/rtp/gstrtpmpapay.c b/gst/rtp/gstrtpmpapay.c
index d4ab883..3962154 100644
--- a/gst/rtp/gstrtpmpapay.c
+++ b/gst/rtp/gstrtpmpapay.c
@@ -133,8 +133,8 @@ gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 {
   gboolean res;
 
-  gst_basertppayload_set_options (payload, "audio", TRUE, "MPA", 90000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "audio", TRUE, "MPA", 90000);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
@@ -236,7 +236,7 @@ gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
     GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts;
     GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
 
-    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf);
+    ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf);
   }
 
   return ret;
@@ -271,7 +271,7 @@ gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   /* if this buffer is going to overflow the packet, flush what we
    * have. */
-  if (gst_basertppayload_is_filled (basepayload,
+  if (gst_base_rtp_payload_is_filled (basepayload,
           packet_len, rtpmpapay->duration + duration)) {
     ret = gst_rtp_mpa_pay_flush (rtpmpapay);
     avail = 0;
diff --git a/gst/rtp/gstrtpmpvpay.c b/gst/rtp/gstrtpmpvpay.c
index 1d32e9a..5b7eee0 100644
--- a/gst/rtp/gstrtpmpvpay.c
+++ b/gst/rtp/gstrtpmpvpay.c
@@ -131,8 +131,8 @@ gst_rtp_mpv_pay_reset (GstRTPMPVPay * pay)
 static gboolean
 gst_rtp_mpv_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 {
-  gst_basertppayload_set_options (payload, "video", FALSE, "MPV", 90000);
-  return gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "video", FALSE, "MPV", 90000);
+  return gst_base_rtp_payload_set_outcaps (payload, NULL);
 }
 
 static gboolean
@@ -216,7 +216,7 @@ gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
 
     GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
 
-    ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
+    ret = gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
   }
 
   return ret;
@@ -264,7 +264,7 @@ gst_rtp_mpv_pay_handle_buffer (GstBaseRTPPayload * basepayload,
   GST_LOG_OBJECT (rtpmpvpay, "available %d, rtp packet length %d", avail,
       packet_len);
 
-  if (gst_basertppayload_is_filled (basepayload,
+  if (gst_base_rtp_payload_is_filled (basepayload,
           packet_len, rtpmpvpay->duration)) {
     ret = gst_rtp_mpv_pay_flush (rtpmpvpay);
   } else {
diff --git a/gst/rtp/gstrtppcmapay.c b/gst/rtp/gstrtppcmapay.c
index 75d12e8..aac1b57 100644
--- a/gst/rtp/gstrtppcmapay.c
+++ b/gst/rtp/gstrtppcmapay.c
@@ -102,8 +102,8 @@ gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 
   payload->pt = GST_RTP_PAYLOAD_PCMA;
 
-  gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "audio", FALSE, "PCMA", 8000);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
diff --git a/gst/rtp/gstrtppcmupay.c b/gst/rtp/gstrtppcmupay.c
index 665fba7..f076ecc 100644
--- a/gst/rtp/gstrtppcmupay.c
+++ b/gst/rtp/gstrtppcmupay.c
@@ -102,8 +102,8 @@ gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
 
   payload->pt = GST_RTP_PAYLOAD_PCMU;
 
-  gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
-  res = gst_basertppayload_set_outcaps (payload, NULL);
+  gst_base_rtp_payload_set_options (payload, "audio", FALSE, "PCMU", 8000);
+  res = gst_base_rtp_payload_set_outcaps (payload, NULL);
 
   return res;
 }
diff --git a/gst/rtp/gstrtpsirenpay.c b/gst/rtp/gstrtpsirenpay.c
index 2aca54a..5a9fc59 100644
--- a/gst/rtp/gstrtpsirenpay.c
+++ b/gst/rtp/gstrtpsirenpay.c
@@ -117,12 +117,12 @@ gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
   if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
     goto wrong_caps;
 
-  gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
+  gst_base_rtp_payload_set_options (basertppayload, "audio", TRUE, "SIREN",
       16000);
   /* set options for this frame based audio codec */
   gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
 
-  return gst_basertppayload_set_outcaps (basertppayload, NULL);
+  return gst_base_rtp_payload_set_outcaps (basertppayload, NULL);
 
   /* ERRORS */
 wrong_dct:
diff --git a/gst/rtp/gstrtpspeexpay.c b/gst/rtp/gstrtpspeexpay.c
index d65f81e..152d4ae 100644
--- a/gst/rtp/gstrtpspeexpay.c
+++ b/gst/rtp/gstrtpspeexpay.c
@@ -184,9 +184,9 @@ gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
 
   payload = GST_BASE_RTP_PAYLOAD (rtpspeexpay);
 
-  gst_basertppayload_set_options (payload, "audio", FALSE, "SPEEX", rate);
+  gst_base_rtp_payload_set_options (payload, "audio", FALSE, "SPEEX", rate);
   cstr = g_strdup_printf ("%d", nb_channels);
-  res = gst_basertppayload_set_outcaps (payload, "encoding-params",
+  res = gst_base_rtp_payload_set_outcaps (payload, "encoding-params",
       G_TYPE_STRING, cstr, NULL);
   g_free (cstr);
 
@@ -290,7 +290,7 @@ gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   gst_rtp_buffer_unmap (&rtp);
 
-  ret = gst_basertppayload_push (basepayload, outbuf);
+  ret = gst_base_rtp_payload_push (basepayload, outbuf);
 
 done:
   gst_buffer_unmap (buffer, data, -1);
diff --git a/gst/rtp/gstrtptheoradepay.c b/gst/rtp/gstrtptheoradepay.c
index a4b98ed..23c6c4e 100644
--- a/gst/rtp/gstrtptheoradepay.c
+++ b/gst/rtp/gstrtptheoradepay.c
@@ -404,7 +404,6 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstFlowReturn ret;
   gint payload_len;
   guint8 *payload, *to_free = NULL;
-  guint32 timestamp;
   guint32 header, ident;
   guint8 F, TDT, packets;
   GstRTPBuffer rtp;
@@ -525,8 +524,6 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
    * ..                        theora data                           |
    * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+*
    */
-  timestamp = gst_rtp_buffer_get_timestamp (&rtp);
-
   while (payload_len >= 2) {
     guint16 length;
 
@@ -572,18 +569,9 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
     payload += length;
     payload_len -= length;
 
-    if (timestamp != -1)
-      /* push with timestamp of the last packet, which is the same timestamp that
-       * should apply to the first assembled packet. */
-      ret = gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
-    else
-      ret = gst_base_rtp_depayload_push (depayload, outbuf);
-
+    ret = gst_base_rtp_depayload_push (depayload, outbuf);
     if (ret != GST_FLOW_OK)
       break;
-
-    /* make sure we don't set a timestamp on next buffers */
-    timestamp = -1;
   }
 
   g_free (to_free);
diff --git a/gst/rtp/gstrtptheorapay.c b/gst/rtp/gstrtptheorapay.c
index 496f57c..1dca934 100644
--- a/gst/rtp/gstrtptheorapay.c
+++ b/gst/rtp/gstrtptheorapay.c
@@ -259,7 +259,7 @@ gst_rtp_theora_pay_flush_packet (GstRtpTheoraPay * rtptheorapay)
 
   /* push, this gives away our ref to the packet, so clear it. */
   ret =
-      gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtptheorapay),
+      gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtptheorapay),
       rtptheorapay->packet);
   rtptheorapay->packet = NULL;
 
@@ -445,13 +445,13 @@ gst_rtp_theora_pay_finish_headers (GstBaseRTPPayload * basepayload)
   /* configure payloader settings */
   wstr = g_strdup_printf ("%d", rtptheorapay->width);
   hstr = g_strdup_printf ("%d", rtptheorapay->height);
-  gst_basertppayload_set_options (basepayload, "video", TRUE, "THEORA", 90000);
-  res = gst_basertppayload_set_outcaps (basepayload,
-      "sampling", G_TYPE_STRING, "YCbCr-4:2:0",
-      "width", G_TYPE_STRING, wstr,
-      "height", G_TYPE_STRING, hstr,
-      "configuration", G_TYPE_STRING, configuration,
-      "delivery-method", G_TYPE_STRING, "inline",
+  gst_base_rtp_payload_set_options (basepayload, "video", TRUE, "THEORA",
+      90000);
+  res =
+      gst_base_rtp_payload_set_outcaps (basepayload, "sampling", G_TYPE_STRING,
+      "YCbCr-4:2:0", "width", G_TYPE_STRING, wstr, "height", G_TYPE_STRING,
+      hstr, "configuration", G_TYPE_STRING, configuration, "delivery-method",
+      G_TYPE_STRING, "inline",
       /* don't set the other defaults 
        */
       NULL);
@@ -550,7 +550,7 @@ gst_rtp_theora_pay_payload_buffer (GstRtpTheoraPay * rtptheorapay, guint8 TDT,
   packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
 
   /* check buffer filled against length and max latency */
-  flush = gst_basertppayload_is_filled (GST_BASE_RTP_PAYLOAD (rtptheorapay),
+  flush = gst_base_rtp_payload_is_filled (GST_BASE_RTP_PAYLOAD (rtptheorapay),
       packet_len, newduration);
   /* we can store up to 15 theora packets in one RTP packet. */
   flush |= (rtptheorapay->payload_pkts == 15);
diff --git a/gst/rtp/gstrtpvorbisdepay.c b/gst/rtp/gstrtpvorbisdepay.c
index e6a2b5b..da4e730 100644
--- a/gst/rtp/gstrtpvorbisdepay.c
+++ b/gst/rtp/gstrtpvorbisdepay.c
@@ -440,7 +440,6 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstFlowReturn ret;
   gint payload_len;
   guint8 *payload, *to_free = NULL;
-  guint32 timestamp;
   guint32 header, ident;
   guint8 F, VDT, packets;
   GstRTPBuffer rtp;
@@ -564,8 +563,6 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
    * ..                        vorbis data                           |
    * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+*
    */
-  timestamp = gst_rtp_buffer_get_timestamp (&rtp);
-
   while (payload_len > 2) {
     guint16 length;
 
@@ -608,18 +605,9 @@ gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
     payload += length;
     payload_len -= length;
 
-    if (timestamp != -1)
-      /* push with timestamp of the last packet, which is the same timestamp that
-       * should apply to the first assembled packet. */
-      ret = gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
-    else
-      ret = gst_base_rtp_depayload_push (depayload, outbuf);
-
+    ret = gst_base_rtp_depayload_push (depayload, outbuf);
     if (ret != GST_FLOW_OK)
       break;
-
-    /* make sure we don't set a timestamp on next buffers */
-    timestamp = -1;
   }
 
   g_free (to_free);
diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c
index 09f71e6..cacaaf1 100644
--- a/gst/rtp/gstrtpvorbispay.c
+++ b/gst/rtp/gstrtpvorbispay.c
@@ -218,7 +218,7 @@ gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
 
   /* push, this gives away our ref to the packet, so clear it. */
   ret =
-      gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
+      gst_base_rtp_payload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
       rtpvorbispay->packet);
   rtpvorbispay->packet = NULL;
 
@@ -385,10 +385,10 @@ gst_rtp_vorbis_pay_finish_headers (GstBaseRTPPayload * basepayload)
 
   /* configure payloader settings */
   cstr = g_strdup_printf ("%d", rtpvorbispay->channels);
-  gst_basertppayload_set_options (basepayload, "audio", TRUE, "VORBIS",
+  gst_base_rtp_payload_set_options (basepayload, "audio", TRUE, "VORBIS",
       rtpvorbispay->rate);
   res =
-      gst_basertppayload_set_outcaps (basepayload, "encoding-params",
+      gst_base_rtp_payload_set_outcaps (basepayload, "encoding-params",
       G_TYPE_STRING, cstr, "configuration", G_TYPE_STRING, configuration, NULL);
   g_free (cstr);
   g_free (configuration);
@@ -543,7 +543,7 @@ gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload,
   packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
 
   /* check buffer filled against length and max latency */
-  flush = gst_basertppayload_is_filled (basepayload, packet_len, newduration);
+  flush = gst_base_rtp_payload_is_filled (basepayload, packet_len, newduration);
   /* we can store up to 15 vorbis packets in one RTP packet. */
   flush |= (rtpvorbispay->payload_pkts == 15);
   /* flush if we have a new VDT */
diff --git a/gst/rtp/gstrtpvrawdepay.c b/gst/rtp/gstrtpvrawdepay.c
index aff0a1b..c3e79ca 100644
--- a/gst/rtp/gstrtpvrawdepay.c
+++ b/gst/rtp/gstrtpvrawdepay.c
@@ -312,8 +312,7 @@ gst_rtp_vraw_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
     GST_LOG_OBJECT (depayload, "new frame with timestamp %u", timestamp);
     /* new timestamp, flush old buffer and create new output buffer */
     if (rtpvrawdepay->outbuf) {
-      gst_base_rtp_depayload_push_ts (depayload, rtpvrawdepay->timestamp,
-          rtpvrawdepay->outbuf);
+      gst_base_rtp_depayload_push (depayload, rtpvrawdepay->outbuf);
       rtpvrawdepay->outbuf = NULL;
     }
 
@@ -526,8 +525,7 @@ gst_rtp_vraw_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   if (gst_rtp_buffer_get_marker (&rtp)) {
     GST_LOG_OBJECT (depayload, "marker, flushing frame");
     if (rtpvrawdepay->outbuf) {
-      gst_base_rtp_depayload_push_ts (depayload, timestamp,
-          rtpvrawdepay->outbuf);
+      gst_base_rtp_depayload_push (depayload, rtpvrawdepay->outbuf);
       rtpvrawdepay->outbuf = NULL;
     }
     rtpvrawdepay->timestamp = -1;
diff --git a/gst/rtp/gstrtpvrawpay.c b/gst/rtp/gstrtpvrawpay.c
index ae6dc83..c2b46af 100644
--- a/gst/rtp/gstrtpvrawpay.c
+++ b/gst/rtp/gstrtpvrawpay.c
@@ -203,14 +203,14 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   wstr = g_strdup_printf ("%d", GST_VIDEO_INFO_WIDTH (&info));
   hstr = g_strdup_printf ("%d", GST_VIDEO_INFO_HEIGHT (&info));
 
-  gst_basertppayload_set_options (payload, "video", TRUE, "RAW", 90000);
+  gst_base_rtp_payload_set_options (payload, "video", TRUE, "RAW", 90000);
   if (info.flags & GST_VIDEO_FLAG_INTERLACED) {
-    res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING,
+    res = gst_base_rtp_payload_set_outcaps (payload, "sampling", G_TYPE_STRING,
         samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING,
         wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING,
         colorimetrystr, "interlace", G_TYPE_STRING, "true", NULL);
   } else {
-    res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING,
+    res = gst_base_rtp_payload_set_outcaps (payload, "sampling", G_TYPE_STRING,
         samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING,
         wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING,
         colorimetrystr, NULL);
@@ -494,7 +494,7 @@ gst_rtp_vraw_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
       }
 
       /* push buffer */
-      ret = gst_basertppayload_push (payload, out);
+      ret = gst_base_rtp_payload_push (payload, out);
     }
 
   }