From: Tim-Philipp Müller <tim@centricular.com>
Date: Wed, 24 May 2017 15:32:30 +0000 (+0100)
Subject: rtpopusdepay: minor perf improvements
X-Git-Tag: 1.19.3~509^2~2062
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=a9f916600474844c2481aebfd6122a3e33422955;p=platform%2Fupstream%2Fgstreamer.git

rtpopusdepay: minor perf improvements

Use the ::process_rtp_packet() vfunc to avoid mapping the
RTP buffer twice.

gst_rtp_buffer_get_payload_buffer() returns a new sub-buffer
which will always be writable, so no need to make it writable.
---

diff --git a/gst/rtp/gstrtpopusdepay.c b/gst/rtp/gstrtpopusdepay.c
index 293371f..f672339 100644
--- a/gst/rtp/gstrtpopusdepay.c
+++ b/gst/rtp/gstrtpopusdepay.c
@@ -52,7 +52,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
     );
 
 static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
-    GstBuffer * buf);
+    GstRTPBuffer * rtp_buffer);
 static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
     GstCaps * caps);
 
@@ -77,7 +77,7 @@ gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
       "Extracts Opus audio from RTP packets",
       "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
 
-  gstbasertpdepayload_class->process = gst_rtp_opus_depay_process;
+  gstbasertpdepayload_class->process_rtp_packet = gst_rtp_opus_depay_process;
   gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps;
 
   GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
@@ -140,16 +140,13 @@ gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
 }
 
 static GstBuffer *
-gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
+gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
+    GstRTPBuffer * rtp_buffer)
 {
   GstBuffer *outbuf;
-  GstRTPBuffer rtpbuf = { NULL, };
 
-  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
-  outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
-  gst_rtp_buffer_unmap (&rtpbuf);
+  outbuf = gst_rtp_buffer_get_payload_buffer (rtp_buffer);
 
-  outbuf = gst_buffer_make_writable (outbuf);
   gst_rtp_drop_non_audio_meta (depayload, outbuf);
 
   return outbuf;