rtp*pay: Allocate using the base class for audio codecs

This is required to add RTP header extensions from the
meta automatically.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>

gst/rtp/gstrtpac3pay.c

@@ -274,7 +274,9 @@ gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay)
     payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
 
     /* create buffer to hold the payload */
-    outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
+    outbuf =
+        gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+        (rtpac3pay), 2, 0, 0);
 
     if (FT == 0) {
       /* check if it all fits */

gst/rtp/gstrtpamrpay.c

@@ -312,7 +312,9 @@ gst_rtp_amr_pay_handle_buffer (GstRTPBasePayload * basepayload,
     goto too_big;
 
   /* now alloc output buffer */
-  outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+  outbuf =
+      gst_rtp_base_payload_allocate_output_buffer (basepayload, payload_len, 0,
+      0);
 
   gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
 

gst/rtp/gstrtpceltpay.c

@@ -332,7 +332,9 @@ gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
       payload_len, GST_TIME_ARGS (rtpceltpay->qduration));
 
   /* get a big enough packet for the sizes + payloads */
-  outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+  outbuf =
+      gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+      (rtpceltpay), payload_len, 0, 0);
 
   GST_BUFFER_DURATION (outbuf) = duration;
 

gst/rtp/gstrtpg723pay.c

@@ -144,8 +144,9 @@ gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
 
   avail = gst_adapter_available (pay->adapter);
 
-  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  outbuf =
-
+      gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (pay),
+      0, 0, 0);
   gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
 
   GST_BUFFER_PTS (outbuf) = pay->timestamp;

gst/rtp/gstrtpg729pay.c

@@ -167,7 +167,9 @@ gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay, GstBuffer * buf)
       payload_len, GST_TIME_ARGS (rtpg729pay->next_ts));
 
   /* create buffer to hold the payload */
-  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  outbuf =
+      gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+      (rtpg729pay), 0, 0, 0);
 
   gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp);
 

gst/rtp/gstrtpgsmpay.c

@@ -145,7 +145,7 @@ gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
   if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
     goto too_big;
 
-  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
   /* copy timestamp and duration */
   GST_BUFFER_PTS (outbuf) = timestamp;

gst/rtp/gstrtpgstpay.c

@@ -299,7 +299,9 @@ gst_rtp_gst_pay_create_from_adapter (GstRtpGSTPay * rtpgstpay,
     payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
 
     /* create buffer to hold the header */
-    outbuf = gst_rtp_buffer_new_allocate (8, 0, 0);
+    outbuf =
+        gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+        (rtpgstpay), 8, 0, 0);
 
     gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
     payload = gst_rtp_buffer_get_payload (&rtp);

gst/rtp/gstrtpmp2tpay.c

@@ -152,7 +152,9 @@ gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
       break;
 
     /* create buffer to hold the payload */
-    outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+    outbuf =
+        gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+        (rtpmp2tpay), 0, 0, 0);
 
     /* get payload */
     paybuf = gst_adapter_take_buffer_fast (rtpmp2tpay->adapter, payload_len);

gst/rtp/gstrtpmp4apay.c

@@ -408,7 +408,8 @@ gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
         packet_len, payload_len);
 
     /* create buffer to hold the payload. */
-    outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0);
+    outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload,
+        header_len, 0, 0);
 
     /* copy payload */
     gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);

gst/rtp/gstrtpmp4gpay.c

@@ -487,8 +487,9 @@ gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
         packet_len, payload_len);
 
     /* create buffer to hold the payload, also make room for the 4 header bytes. */
-    outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
+    outbuf =
-
+        gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+        (rtpmp4gpay), 4, 0, 0);
     gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
 
     /* copy payload */

gst/rtp/gstrtpmpapay.c

@@ -213,7 +213,9 @@ gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
     payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
 
     /* create buffer to hold the payload */
-    outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
+    outbuf =
+        gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+        (rtpmpapay), 4, 0, 0);
 
     gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
 

gst/rtp/gstrtpsbcpay.c

@@ -198,7 +198,9 @@ gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
     if (payload_length == 0)    /* Nothing to send */
       return GST_FLOW_OK;
 
-    outbuf = gst_rtp_buffer_new_allocate (RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
+    outbuf =
+        gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+        (sbcpay), RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
 
     /* get payload */
     gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);

gst/rtp/gstrtpspeexpay.c

@@ -279,8 +279,8 @@ gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
   duration = GST_BUFFER_DURATION (buffer);
 
   /* FIXME, only one SPEEX frame per RTP packet for now */
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
-  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
   /* FIXME, assert for now */
   g_assert (gst_buffer_get_size (buffer) <=
       GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay));

gst/rtp/gstrtpvorbispay.c

@@ -270,14 +270,18 @@ static void
 gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT,
     GstClockTime timestamp)
 {
+  guint len;
+
   GST_LOG_OBJECT (rtpvorbispay, "starting new packet, VDT: %d", VDT);
 
   gst_rtp_vorbis_pay_clear_packet (rtpvorbispay);
 
   /* new packet allocate max packet size */
-  rtpvorbispay->packet =
+  len = gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU
-      gst_rtp_buffer_new_allocate_len (GST_RTP_BASE_PAYLOAD_MTU
       (rtpvorbispay), 0, 0);
+  rtpvorbispay->packet =
+      gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+      (rtpvorbispay), len, 0, 0);
   gst_rtp_vorbis_pay_reset_packet (rtpvorbispay, VDT);
 
   GST_BUFFER_PTS (rtpvorbispay->packet) = timestamp;