rtp: port more to 0.11

diff --git a/gst/rtp/gstrtpg722depay.c b/gst/rtp/gstrtpg722depay.c
index 1e892fb..8f0f2c5 100644 (file)
--- a/gst/rtp/gstrtpg722depay.c
+++ b/gst/rtp/gstrtpg722depay.c
@@ -62,7 +62,8 @@ static GstStaticPadTemplate gst_rtp_g722_depay_sink_template =
     )
     );
 
-GST_BOILERPLATE (GstRtpG722Depay, gst_rtp_g722_depay, GstBaseRTPDepayload,
+#define gst_rtp_g722_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpG722Depay, gst_rtp_g722_depay,
     GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static gboolean gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload,
@@ -71,40 +72,34 @@ static GstBuffer *gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload,
     GstBuffer * buf);
 
 static void
-gst_rtp_g722_depay_base_init (gpointer klass)
+gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
+      "G722 RTP Depayloader");
 
-  gst_element_class_add_pad_template (element_class,
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g722_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g722_depay_sink_template));
 
-  gst_element_class_set_details_simple (element_class, "RTP audio depayloader",
-      "Codec/Depayloader/Network/RTP",
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP audio depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts G722 audio from RTP packets",
       "Wim Taymans <wim.taymans@gmail.com>");
-}
-
-static void
-gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
-{
-  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
-  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
   gstbasertpdepayload_class->set_caps = gst_rtp_g722_depay_setcaps;
   gstbasertpdepayload_class->process = gst_rtp_g722_depay_process;
-
-  GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
-      "G722 RTP Depayloader");
 }
 
 static void
-gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay,
-    GstRtpG722DepayClass * klass)
+gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay)
 {
-  /* needed because of GST_BOILERPLATE */
 }
 
 static gint
@@ -223,18 +218,22 @@ gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstBuffer *outbuf;
   gint payload_len;
   gboolean marker;
+  GstRTPBuffer rtp = { NULL };
 
   rtpg722depay = GST_RTP_G722_DEPAY (depayload);
 
-  payload_len = gst_rtp_buffer_get_payload_len (buf);
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+
+  payload_len = gst_rtp_buffer_get_payload_len (&rtp);
 
   if (payload_len <= 0)
     goto empty_packet;
 
   GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
 
-  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
-  marker = gst_rtp_buffer_get_marker (buf);
+  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+  marker = gst_rtp_buffer_get_marker (&rtp);
+  gst_rtp_buffer_unmap (&rtp);
 
   if (marker) {
     /* mark talk spurt with DISCONT */
@@ -248,6 +247,7 @@ empty_packet:
   {
     GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
         ("Empty Payload."), (NULL));
+    gst_rtp_buffer_unmap (&rtp);
     return NULL;
   }
 }

diff --git a/gst/rtp/gstrtpg722pay.c b/gst/rtp/gstrtpg722pay.c
index 29cd22a..b2c8ff4 100644 (file)
--- a/gst/rtp/gstrtpg722pay.c
+++ b/gst/rtp/gstrtpg722pay.c
@@ -56,41 +56,38 @@ static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload,
 static GstCaps *gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload,
     GstPad * pad);
 
-GST_BOILERPLATE (GstRtpG722Pay, gst_rtp_g722_pay, GstBaseRTPAudioPayload,
+#define gst_rtp_g722_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpG722Pay, gst_rtp_g722_pay,
     GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
 
 static void
-gst_rtp_g722_pay_base_init (gpointer klass)
+gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
+      "G722 RTP Payloader");
 
-  gst_element_class_add_pad_template (element_class,
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g722_pay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template));
 
-  gst_element_class_set_details_simple (element_class, "RTP audio payloader",
+  gst_element_class_set_details_simple (gstelement_class, "RTP audio payloader",
       "Codec/Payloader/Network/RTP",
       "Payload-encode Raw audio into RTP packets (RFC 3551)",
       "Wim Taymans <wim.taymans@gmail.com>");
-}
-
-static void
-gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
-{
-  GstBaseRTPPayloadClass *gstbasertppayload_class;
-
-  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   gstbasertppayload_class->set_caps = gst_rtp_g722_pay_setcaps;
   gstbasertppayload_class->get_caps = gst_rtp_g722_pay_getcaps;
-
-  GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
-      "G722 RTP Payloader");
 }
 
 static void
-gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass)
+gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay)
 {
   GstBaseRTPAudioPayload *basertpaudiopayload;
 

diff --git a/gst/rtp/gstrtpg723depay.c b/gst/rtp/gstrtpg723depay.c
index b4aa5fe..0823ae7 100644 (file)
--- a/gst/rtp/gstrtpg723depay.c
+++ b/gst/rtp/gstrtpg723depay.c
@@ -78,42 +78,38 @@ static gboolean gst_rtp_g723_depay_setcaps (GstBaseRTPDepayload * depayload,
 static GstBuffer *gst_rtp_g723_depay_process (GstBaseRTPDepayload * depayload,
     GstBuffer * buf);
 
-GST_BOILERPLATE (GstRtpG723Depay, gst_rtp_g723_depay, GstBaseRTPDepayload,
+#define gst_rtp_g723_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpG723Depay, gst_rtp_g723_depay,
     GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static void
-gst_rtp_g723_depay_base_init (gpointer klass)
+gst_rtp_g723_depay_class_init (GstRtpG723DepayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  GST_DEBUG_CATEGORY_INIT (rtpg723depay_debug, "rtpg723depay", 0,
+      "G.723 RTP Depayloader");
 
-  gst_element_class_add_pad_template (element_class,
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g723_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g723_depay_sink_template));
 
-  gst_element_class_set_details_simple (element_class, "RTP G.723 depayloader",
-      "Codec/Depayloader/Network/RTP",
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP G.723 depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts G.723 audio from RTP packets (RFC 3551)",
       "Wim Taymans <wim.taymans@gmail.com>");
 
-  GST_DEBUG_CATEGORY_INIT (rtpg723depay_debug, "rtpg723depay", 0,
-      "G.723 RTP Depayloader");
-}
-
-static void
-gst_rtp_g723_depay_class_init (GstRtpG723DepayClass * klass)
-{
-  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
-  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
-
   gstbasertpdepayload_class->process = gst_rtp_g723_depay_process;
   gstbasertpdepayload_class->set_caps = gst_rtp_g723_depay_setcaps;
 }
 
 static void
-gst_rtp_g723_depay_init (GstRtpG723Depay * rtpg723depay,
-    GstRtpG723DepayClass * klass)
+gst_rtp_g723_depay_init (GstRtpG723Depay * rtpg723depay)
 {
   GstBaseRTPDepayload *depayload;
 
@@ -182,10 +178,13 @@ gst_rtp_g723_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstBuffer *outbuf = NULL;
   gint payload_len;
   gboolean marker;
+  GstRTPBuffer rtp = { NULL };
 
   rtpg723depay = GST_RTP_G723_DEPAY (depayload);
 
-  payload_len = gst_rtp_buffer_get_payload_len (buf);
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+
+  payload_len = gst_rtp_buffer_get_payload_len (&rtp);
 
   /* At least 4 bytes */
   if (payload_len < 4)
@@ -193,8 +192,9 @@ gst_rtp_g723_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
 
   GST_LOG_OBJECT (rtpg723depay, "payload len %d", payload_len);
 
-  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
-  marker = gst_rtp_buffer_get_marker (buf);
+  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+  marker = gst_rtp_buffer_get_marker (&rtp);
+  gst_rtp_buffer_unmap (&rtp);
 
   if (marker) {
     /* marker bit starts talkspurt */
@@ -202,7 +202,7 @@ gst_rtp_g723_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   }
 
   GST_LOG_OBJECT (depayload, "pushing buffer of size %d",
-      GST_BUFFER_SIZE (outbuf));
+      gst_buffer_get_size (outbuf));
 
   return outbuf;
 
@@ -216,6 +216,7 @@ too_small:
 bad_packet:
   {
     /* no fatal error */
+    gst_rtp_buffer_unmap (&rtp);
     return NULL;
   }
 }

diff --git a/gst/rtp/gstrtpg723pay.c b/gst/rtp/gstrtpg723pay.c
index 439ef02..2621c55 100644 (file)
--- a/gst/rtp/gstrtpg723pay.c
+++ b/gst/rtp/gstrtpg723pay.c
@@ -67,23 +67,8 @@ static void gst_rtp_g723_pay_finalize (GObject * object);
 static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
     GstStateChange transition);
 
-GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPPayload,
-    GST_TYPE_BASE_RTP_PAYLOAD);
-
-static void
-gst_rtp_g723_pay_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g723_pay_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g723_pay_src_template));
-  gst_element_class_set_details_simple (element_class, "RTP G.723 payloader",
-      "Codec/Payloader/Network/RTP",
-      "Packetize G.723 audio into RTP packets",
-      "Wim Taymans <wim.taymans@gmail.com>");
-}
+#define gst_rtp_g723_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRTPG723Pay, gst_rtp_g723_pay, GST_TYPE_BASE_RTP_PAYLOAD);
 
 static void
 gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
@@ -100,12 +85,22 @@ gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
 
   gstelement_class->change_state = gst_rtp_g723_pay_change_state;
 
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g723_pay_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g723_pay_src_template));
+
+  gst_element_class_set_details_simple (gstelement_class, "RTP G.723 payloader",
+      "Codec/Payloader/Network/RTP",
+      "Packetize G.723 audio into RTP packets",
+      "Wim Taymans <wim.taymans@gmail.com>");
+
   payload_class->set_caps = gst_rtp_g723_pay_set_caps;
   payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
 }
 
 static void
-gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass)
+gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
 {
   GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
 
@@ -155,11 +150,14 @@ gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
   GstFlowReturn ret;
   guint8 *payload;
   guint avail;
+  GstRTPBuffer rtp = { NULL };
 
   avail = gst_adapter_available (pay->adapter);
 
   outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
-  payload = gst_rtp_buffer_get_payload (outbuf);
+
+  gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
+  payload = gst_rtp_buffer_get_payload (&rtp);
 
   GST_BUFFER_TIMESTAMP (outbuf) = pay->timestamp;
   GST_BUFFER_DURATION (outbuf) = pay->duration;
@@ -175,9 +173,10 @@ gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
   /* set discont and marker */
   if (pay->discont) {
     GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
-    gst_rtp_buffer_set_marker (outbuf, TRUE);
+    gst_rtp_buffer_set_marker (&rtp, TRUE);
     pay->discont = FALSE;
   }
+  gst_rtp_buffer_unmap (&rtp);
 
   ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
 
@@ -197,7 +196,7 @@ gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
 {
   GstFlowReturn ret = GST_FLOW_OK;
   guint8 *data;
-  guint size;
+  gsize size;
   guint8 HDR;
   GstRTPG723Pay *pay;
   GstClockTime packet_dur, timestamp;
@@ -205,8 +204,7 @@ gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
 
   pay = GST_RTP_G723_PAY (payload);
 
-  size = GST_BUFFER_SIZE (buf);
-  data = GST_BUFFER_DATA (buf);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
   timestamp = GST_BUFFER_TIMESTAMP (buf);
 
   if (GST_BUFFER_IS_DISCONT (buf)) {
@@ -244,6 +242,7 @@ gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
     else
       pay->timestamp = 0;
   }
+  gst_buffer_unmap (buf, data, size);
 
   /* add packet to the queue */
   gst_adapter_push (pay->adapter, buf);
@@ -262,6 +261,7 @@ invalid_size:
     GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
         ("Invalid input buffer size"),
         ("Input size should be 4, 20 or 24, got %u", size));
+    gst_buffer_unmap (buf, data, size);
     gst_buffer_unref (buf);
     return GST_FLOW_OK;
   }
@@ -270,6 +270,7 @@ wrong_size:
     GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
         ("Wrong input buffer size"),
         ("Expected input buffer size %u but got %u", size_tab[HDR], size));
+    gst_buffer_unmap (buf, data, size);
     gst_buffer_unref (buf);
     return GST_FLOW_OK;
   }

diff --git a/gst/rtp/gstrtpg726depay.c b/gst/rtp/gstrtpg726depay.c
index 2b36755..cf88bd6 100644 (file)
--- a/gst/rtp/gstrtpg726depay.c
+++ b/gst/rtp/gstrtpg726depay.c
@@ -87,31 +87,22 @@ static GstBuffer *gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload,
 static gboolean gst_rtp_g726_depay_setcaps (GstBaseRTPDepayload * depayload,
     GstCaps * caps);
 
-GST_BOILERPLATE (GstRtpG726Depay, gst_rtp_g726_depay, GstBaseRTPDepayload,
+#define gst_rtp_g726_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpG726Depay, gst_rtp_g726_depay,
     GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static void
-gst_rtp_g726_depay_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g726_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g726_depay_sink_template));
-  gst_element_class_set_details_simple (element_class, "RTP G.726 depayloader",
-      "Codec/Depayloader/Network/RTP",
-      "Extracts G.726 audio from RTP packets",
-      "Axis Communications <dev-gstreamer@axis.com>");
-}
-
-static void
 gst_rtp_g726_depay_class_init (GstRtpG726DepayClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
   GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
 
+  GST_DEBUG_CATEGORY_INIT (rtpg726depay_debug, "rtpg726depay", 0,
+      "G.726 RTP Depayloader");
+
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
   gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
   gobject_class->set_property = gst_rtp_g726_depay_set_property;
@@ -122,16 +113,22 @@ gst_rtp_g726_depay_class_init (GstRtpG726DepayClass * klass)
           "Force AAL2 decoding for compatibility with bad payloaders",
           DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g726_depay_src_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g726_depay_sink_template));
+
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP G.726 depayloader", "Codec/Depayloader/Network/RTP",
+      "Extracts G.726 audio from RTP packets",
+      "Axis Communications <dev-gstreamer@axis.com>");
+
   gstbasertpdepayload_class->process = gst_rtp_g726_depay_process;
   gstbasertpdepayload_class->set_caps = gst_rtp_g726_depay_setcaps;
-
-  GST_DEBUG_CATEGORY_INIT (rtpg726depay_debug, "rtpg726depay", 0,
-      "G.726 RTP Depayloader");
 }
 
 static void
-gst_rtp_g726_depay_init (GstRtpG726Depay * rtpG726depay,
-    GstRtpG726DepayClass * klass)
+gst_rtp_g726_depay_init (GstRtpG726Depay * rtpG726depay)
 {
   GstBaseRTPDepayload *depayload;
 
@@ -210,36 +207,34 @@ gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstRtpG726Depay *depay;
   GstBuffer *outbuf = NULL;
   gboolean marker;
+  GstRTPBuffer rtp = { NULL };
 
   depay = GST_RTP_G726_DEPAY (depayload);
 
-  marker = gst_rtp_buffer_get_marker (buf);
+  gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
+
+  marker = gst_rtp_buffer_get_marker (&rtp);
 
   GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
-      GST_BUFFER_SIZE (buf), marker,
-      gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
+      gst_buffer_get_size (buf), marker,
+      gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
 
   if (depay->aal2 || depay->force_aal2) {
     /* AAL2, we can just copy the bytes */
-    outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+    outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
   } else {
-    guint8 *in, *out, tmp;
+    guint8 *in, *out, tmp, *odata;
     guint len;
+    gsize osize;
 
-    in = gst_rtp_buffer_get_payload (buf);
-    len = gst_rtp_buffer_get_payload_len (buf);
+    in = gst_rtp_buffer_get_payload (&rtp);
+    len = gst_rtp_buffer_get_payload_len (&rtp);
 
-    if (gst_buffer_is_writable (buf)) {
-      outbuf = gst_rtp_buffer_get_payload_buffer (buf);
-    } else {
-      GstBuffer *copy;
+    outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+    outbuf = gst_buffer_make_writable (outbuf);
 
-      /* copy buffer */
-      copy = gst_buffer_copy (buf);
-      outbuf = gst_rtp_buffer_get_payload_buffer (copy);
-      gst_buffer_unref (copy);
-    }
-    out = GST_BUFFER_DATA (outbuf);
+    odata = gst_buffer_map (outbuf, &osize, NULL, GST_MAP_WRITE);
+    out = odata;
 
     /* we need to reshuffle the bytes, input is always of the form
      * A B C D ... with the number of bits depending on the bitrate. */
@@ -327,6 +322,7 @@ gst_rtp_g726_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
         break;
       }
     }
+    gst_buffer_unmap (outbuf, odata, osize);
   }
 
   if (marker) {

diff --git a/gst/rtp/gstrtpg726pay.c b/gst/rtp/gstrtpg726pay.c
index 16086bf..71ac107 100644 (file)
--- a/gst/rtp/gstrtpg726pay.c
+++ b/gst/rtp/gstrtpg726pay.c
@@ -75,31 +75,19 @@ static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
 static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload *
     payload, GstBuffer * buffer);
 
-GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
+#define gst_rtp_g726_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpG726Pay, gst_rtp_g726_pay,
     GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
 
 static void
-gst_rtp_g726_pay_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g726_pay_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
-  gst_element_class_set_details_simple (element_class, "RTP G.726 payloader",
-      "Codec/Payloader/Network/RTP",
-      "Payload-encodes G.726 audio into a RTP packet",
-      "Axis Communications <dev-gstreamer@axis.com>");
-}
-
-static void
 gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
   GstBaseRTPPayloadClass *gstbasertppayload_class;
 
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
   gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   gobject_class->set_property = gst_rtp_g726_pay_set_property;
@@ -110,6 +98,16 @@ gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
           "Force AAL2 encoding for compatibility with bad depayloaders",
           DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g726_pay_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
+
+  gst_element_class_set_details_simple (gstelement_class, "RTP G.726 payloader",
+      "Codec/Payloader/Network/RTP",
+      "Payload-encodes G.726 audio into a RTP packet",
+      "Axis Communications <dev-gstreamer@axis.com>");
+
   gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer;
 
@@ -118,7 +116,7 @@ gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
 }
 
 static void
-gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
+gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay)
 {
   GstBaseRTPAudioPayload *basertpaudiopayload;
 
@@ -269,14 +267,13 @@ gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
 
   if (!pay->aal2) {
     guint8 *data, tmp;
-    guint len;
+    gsize len;
 
     /* for non AAL2, we need to reshuffle the bytes, we can do this in-place
      * when the buffer is writable. */
     buffer = gst_buffer_make_writable (buffer);
 
-    data = GST_BUFFER_DATA (buffer);
-    len = GST_BUFFER_SIZE (buffer);
+    data = gst_buffer_map (buffer, &len, NULL, GST_MAP_READWRITE);
 
     GST_LOG_OBJECT (pay, "packing %u bytes of data", len);
 
@@ -366,6 +363,7 @@ gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
         break;
       }
     }
+    gst_buffer_unmap (buffer, data, len);
   }
 
   res =

diff --git a/gst/rtp/gstrtpg729depay.c b/gst/rtp/gstrtpg729depay.c
index 3d22508..4a9b4af 100644 (file)
--- a/gst/rtp/gstrtpg729depay.c
+++ b/gst/rtp/gstrtpg729depay.c
@@ -76,42 +76,38 @@ static gboolean gst_rtp_g729_depay_setcaps (GstBaseRTPDepayload * depayload,
 static GstBuffer *gst_rtp_g729_depay_process (GstBaseRTPDepayload * depayload,
     GstBuffer * buf);
 
-GST_BOILERPLATE (GstRtpG729Depay, gst_rtp_g729_depay, GstBaseRTPDepayload,
+#define gst_rtp_g729_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpG729Depay, gst_rtp_g729_depay,
     GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static void
-gst_rtp_g729_depay_base_init (gpointer klass)
+gst_rtp_g729_depay_class_init (GstRtpG729DepayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  GST_DEBUG_CATEGORY_INIT (rtpg729depay_debug, "rtpg729depay", 0,
+      "G.729 RTP Depayloader");
 
-  gst_element_class_add_pad_template (element_class,
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g729_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_g729_depay_sink_template));
 
-  gst_element_class_set_details_simple (element_class, "RTP G.729 depayloader",
-      "Codec/Depayloader/Network/RTP",
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP G.729 depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts G.729 audio from RTP packets (RFC 3551)",
       "Laurent Glayal <spglegle@yahoo.fr>");
 
-  GST_DEBUG_CATEGORY_INIT (rtpg729depay_debug, "rtpg729depay", 0,
-      "G.729 RTP Depayloader");
-}
-
-static void
-gst_rtp_g729_depay_class_init (GstRtpG729DepayClass * klass)
-{
-  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
-  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
-
   gstbasertpdepayload_class->process = gst_rtp_g729_depay_process;
   gstbasertpdepayload_class->set_caps = gst_rtp_g729_depay_setcaps;
 }
 
 static void
-gst_rtp_g729_depay_init (GstRtpG729Depay * rtpg729depay,
-    GstRtpG729DepayClass * klass)
+gst_rtp_g729_depay_init (GstRtpG729Depay * rtpg729depay)
 {
   GstBaseRTPDepayload *depayload;
 
@@ -172,7 +168,6 @@ wrong_clock_rate:
   }
 }
 
-
 static GstBuffer *
 gst_rtp_g729_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
 {
@@ -180,10 +175,13 @@ gst_rtp_g729_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstBuffer *outbuf = NULL;
   gint payload_len;
   gboolean marker;
+  GstRTPBuffer rtp = { NULL };
 
   rtpg729depay = GST_RTP_G729_DEPAY (depayload);
 
-  payload_len = gst_rtp_buffer_get_payload_len (buf);
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+
+  payload_len = gst_rtp_buffer_get_payload_len (&rtp);
 
   /* At least 2 bytes (CNG from G729 Annex B) */
   if (payload_len < 2) {
@@ -198,8 +196,10 @@ gst_rtp_g729_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
     GST_LOG_OBJECT (rtpg729depay, "G729 payload contains CNG frame");
   }
 
-  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
-  marker = gst_rtp_buffer_get_marker (buf);
+  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+  marker = gst_rtp_buffer_get_marker (&rtp);
+
+  gst_rtp_buffer_unmap (&rtp);
 
   if (marker) {
     /* marker bit starts talkspurt */
@@ -207,7 +207,7 @@ gst_rtp_g729_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   }
 
   GST_LOG_OBJECT (depayload, "pushing buffer of size %d",
-      GST_BUFFER_SIZE (outbuf));
+      gst_buffer_get_size (outbuf));
 
   return outbuf;
 

diff --git a/gst/rtp/gstrtpg729pay.c b/gst/rtp/gstrtpg729pay.c
index 37049d6..746b2f9 100644 (file)
--- a/gst/rtp/gstrtpg729pay.c
+++ b/gst/rtp/gstrtpg729pay.c
@@ -74,26 +74,8 @@ static GstStaticPadTemplate gst_rtp_g729_pay_src_template =
         "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
     );
 
-GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPPayload,
-    GST_TYPE_BASE_RTP_PAYLOAD);
-
-static void
-gst_rtp_g729_pay_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g729_pay_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_g729_pay_src_template));
-  gst_element_class_set_details_simple (element_class, "RTP G.729 payloader",
-      "Codec/Payloader/Network/RTP",
-      "Packetize G.729 audio into RTP packets",
-      "Olivier Crete <olivier.crete@collabora.co.uk>");
-
-  GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
-      "G.729 RTP Payloader");
-}
+#define gst_rtp_g729_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRTPG729Pay, gst_rtp_g729_pay, GST_TYPE_BASE_RTP_PAYLOAD);
 
 static void
 gst_rtp_g729_pay_finalize (GObject * object)
@@ -112,16 +94,29 @@ gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
   GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
 
+  GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
+      "G.729 RTP Payloader");
+
   gobject_class->finalize = gst_rtp_g729_pay_finalize;
 
   gstelement_class->change_state = gst_rtp_g729_pay_change_state;
 
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g729_pay_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_g729_pay_src_template));
+
+  gst_element_class_set_details_simple (gstelement_class, "RTP G.729 payloader",
+      "Codec/Payloader/Network/RTP",
+      "Packetize G.729 audio into RTP packets",
+      "Olivier Crete <olivier.crete@collabora.co.uk>");
+
   payload_class->set_caps = gst_rtp_g729_pay_set_caps;
   payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
 }
 
 static void
-gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass)
+gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
 {
   GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
 
@@ -170,6 +165,7 @@ gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay,
   GstBuffer *outbuf;
   guint8 *payload;
   GstFlowReturn ret;
+  GstRTPBuffer rtp = { NULL };
 
   basepayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
 
@@ -179,8 +175,10 @@ gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay,
   /* create buffer to hold the payload */
   outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
 
+  gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp);
+
   /* copy payload */
-  payload = gst_rtp_buffer_get_payload (outbuf);
+  payload = gst_rtp_buffer_get_payload (&rtp);
   memcpy (payload, data, payload_len);
 
   /* set metadata */
@@ -196,9 +194,10 @@ gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay,
   if (G_UNLIKELY (rtpg729pay->discont)) {
     GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
     GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
-    gst_rtp_buffer_set_marker (outbuf, TRUE);
+    gst_rtp_buffer_set_marker (&rtp, TRUE);
     rtpg729pay->discont = FALSE;
   }
+  gst_rtp_buffer_unmap (&rtp);
 
   ret = gst_basertppayload_push (basepayload, outbuf);
 
@@ -235,11 +234,13 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
   guint minptime_octets = 0;
   guint min_payload_len;
   guint max_payload_len;
+  gsize size;
+  GstClockTime timestamp;
 
-  available = GST_BUFFER_SIZE (buf);
+  size = gst_buffer_get_size (buf);
 
-  if (available % G729_FRAME_SIZE != 0 &&
-      available % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
+  if (size % G729_FRAME_SIZE != 0 &&
+      size % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
     goto invalid_size;
 
   /* max number of bytes based on given ptime, has to be multiple of
@@ -302,6 +303,8 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
   adapter = rtpg729pay->adapter;
   available = gst_adapter_available (adapter);
 
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+
   /* resync rtp time on discont or a discontinuous cn packet */
   if (GST_BUFFER_IS_DISCONT (buf)) {
     /* flush remainder */
@@ -311,26 +314,27 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
       available = 0;
     }
     rtpg729pay->discont = TRUE;
-    gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, GST_BUFFER_TIMESTAMP (buf));
+    gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
   }
 
-  if (GST_BUFFER_SIZE (buf) < G729_FRAME_SIZE)
-    gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, GST_BUFFER_TIMESTAMP (buf));
+  if (size < G729_FRAME_SIZE)
+    gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
 
   if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts))) {
-    rtpg729pay->first_ts = GST_BUFFER_TIMESTAMP (buf);
+    rtpg729pay->first_ts = timestamp;
     rtpg729pay->first_rtp_time = rtpg729pay->next_rtp_time;
   }
 
   /* let's reset the base timestamp when the adapter is empty */
   if (available == 0)
-    rtpg729pay->next_ts = GST_BUFFER_TIMESTAMP (buf);
+    rtpg729pay->next_ts = timestamp;
 
-  if (available == 0 &&
-      GST_BUFFER_SIZE (buf) >= min_payload_len &&
-      GST_BUFFER_SIZE (buf) <= max_payload_len) {
-    ret = gst_rtp_g729_pay_push (rtpg729pay,
-        GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+  if (available == 0 && size >= min_payload_len && size <= max_payload_len) {
+    guint8 *data;
+
+    data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
+    ret = gst_rtp_g729_pay_push (rtpg729pay, data, size);
+    gst_buffer_unmap (buf, data, size);
     gst_buffer_unref (buf);
     return ret;
   }
@@ -364,7 +368,7 @@ invalid_size:
         ("Invalid input buffer size"),
         ("Invalid buffer size, should be a multiple of"
             " G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)"
-            " added to it, but it is %u", available));
+            " added to it, but it is %u", size));
     gst_buffer_unref (buf);
     return GST_FLOW_ERROR;
   }

diff --git a/gst/rtp/gstrtpgsmdepay.c b/gst/rtp/gstrtpgsmdepay.c
index bb62c50..30f6a7f 100644 (file)
--- a/gst/rtp/gstrtpgsmdepay.c
+++ b/gst/rtp/gstrtpgsmdepay.c
@@ -62,29 +62,26 @@ static GstBuffer *gst_rtp_gsm_depay_process (GstBaseRTPDepayload * _depayload,
 static gboolean gst_rtp_gsm_depay_setcaps (GstBaseRTPDepayload * _depayload,
     GstCaps * caps);
 
-GST_BOILERPLATE (GstRTPGSMDepay, gst_rtp_gsm_depay, GstBaseRTPDepayload,
-    GST_TYPE_BASE_RTP_DEPAYLOAD);
+#define gst_rtp_gsm_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRTPGSMDepay, gst_rtp_gsm_depay, GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static void
-gst_rtp_gsm_depay_base_init (gpointer klass)
+gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertp_depayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertp_depayload_class = (GstBaseRTPDepayloadClass *) klass;
 
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_gsm_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_gsm_depay_sink_template));
-  gst_element_class_set_details_simple (element_class, "RTP GSM depayloader",
+
+  gst_element_class_set_details_simple (gstelement_class, "RTP GSM depayloader",
       "Codec/Depayloader/Network/RTP",
       "Extracts GSM audio from RTP packets", "Zeeshan Ali <zeenix@gmail.com>");
-}
-
-static void
-gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass)
-{
-  GstBaseRTPDepayloadClass *gstbasertp_depayload_class;
-
-  gstbasertp_depayload_class = (GstBaseRTPDepayloadClass *) klass;
 
   gstbasertp_depayload_class->process = gst_rtp_gsm_depay_process;
   gstbasertp_depayload_class->set_caps = gst_rtp_gsm_depay_setcaps;
@@ -94,10 +91,8 @@ gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass)
 }
 
 static void
-gst_rtp_gsm_depay_init (GstRTPGSMDepay * rtpgsmdepay,
-    GstRTPGSMDepayClass * klass)
+gst_rtp_gsm_depay_init (GstRTPGSMDepay * rtpgsmdepay)
 {
-  /* needed because of GST_BOILERPLATE */
 }
 
 static gboolean
@@ -127,14 +122,19 @@ gst_rtp_gsm_depay_process (GstBaseRTPDepayload * _depayload, GstBuffer * buf)
 {
   GstBuffer *outbuf = NULL;
   gboolean marker;
+  GstRTPBuffer rtp = { NULL };
+
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
 
-  marker = gst_rtp_buffer_get_marker (buf);
+  marker = gst_rtp_buffer_get_marker (&rtp);
 
   GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
-      GST_BUFFER_SIZE (buf), marker,
-      gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
+      gst_buffer_get_size (buf), marker,
+      gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
+
+  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
 
-  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+  gst_rtp_buffer_unmap (&rtp);
 
   if (marker) {
     /* mark start of talkspurt with DISCONT */

diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c
index 479013e..5a8061f 100644 (file)
--- a/gst/rtp/gstrtpgsmpay.c
+++ b/gst/rtp/gstrtpgsmpay.c
@@ -57,40 +57,37 @@ static gboolean gst_rtp_gsm_pay_setcaps (GstBaseRTPPayload * payload,
 static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * payload,
     GstBuffer * buffer);
 
-GST_BOILERPLATE (GstRTPGSMPay, gst_rtp_gsm_pay, GstBaseRTPPayload,
-    GST_TYPE_BASE_RTP_PAYLOAD);
+#define gst_rtp_gsm_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_BASE_RTP_PAYLOAD);
 
 static void
-gst_rtp_gsm_pay_base_init (gpointer klass)
+gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
+      "GSM Audio RTP Payloader");
 
-  gst_element_class_add_pad_template (element_class,
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_gsm_pay_sink_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template));
-  gst_element_class_set_details_simple (element_class, "RTP GSM payloader",
+
+  gst_element_class_set_details_simple (gstelement_class, "RTP GSM payloader",
       "Codec/Payloader/Network/RTP",
       "Payload-encodes GSM audio into a RTP packet",
       "Zeeshan Ali <zeenix@gmail.com>");
-}
-
-static void
-gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
-{
-  GstBaseRTPPayloadClass *gstbasertppayload_class;
-
-  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   gstbasertppayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
-
-  GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
-      "GSM Audio RTP Payloader");
 }
 
 static void
-gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay, GstRTPGSMPayClass * klass)
+gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
 {
   GST_BASE_RTP_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
   GST_BASE_RTP_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
@@ -128,15 +125,18 @@ gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * basepayload,
     GstBuffer * buffer)
 {
   GstRTPGSMPay *rtpgsmpay;
-  guint size, payload_len;
+  guint payload_len;
   GstBuffer *outbuf;
   guint8 *payload, *data;
   GstClockTime timestamp, duration;
   GstFlowReturn ret;
+  gsize size;
+  GstRTPBuffer rtp = { NULL };
 
   rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
 
-  size = GST_BUFFER_SIZE (buffer);
+  data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
+
   timestamp = GST_BUFFER_TIMESTAMP (buffer);
   duration = GST_BUFFER_DURATION (buffer);
 
@@ -144,12 +144,8 @@ gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * basepayload,
   payload_len = size;
 
   /* FIXME, just error out for now */
-  if (payload_len > GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay)) {
-    GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
-        ("payload_len %u > mtu %u", payload_len,
-            GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay)));
-    return GST_FLOW_ERROR;
-  }
+  if (payload_len > GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay))
+    goto too_big;
 
   outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
 
@@ -158,21 +154,33 @@ gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * basepayload,
   GST_BUFFER_DURATION (outbuf) = duration;
 
   /* get payload */
-  payload = gst_rtp_buffer_get_payload (outbuf);
-
-  data = GST_BUFFER_DATA (buffer);
+  gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
 
   /* copy data in payload */
-  memcpy (&payload[0], data, size);
+  payload = gst_rtp_buffer_get_payload (&rtp);
+  memcpy (payload, data, size);
+
+  gst_rtp_buffer_unmap (&rtp);
 
+  gst_buffer_unmap (buffer, data, size);
   gst_buffer_unref (buffer);
 
   GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %d",
-      GST_BUFFER_SIZE (outbuf));
+      gst_buffer_get_size (outbuf));
 
   ret = gst_basertppayload_push (basepayload, outbuf);
 
   return ret;
+
+  /* ERRORS */
+too_big:
+  {
+    GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
+        ("payload_len %u > mtu %u", payload_len,
+            GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay)));
+    gst_buffer_unmap (buffer, data, size);
+    return GST_FLOW_ERROR;
+  }
 }
 
 gboolean

diff --git a/gst/rtp/gstrtph263pay.c b/gst/rtp/gstrtph263pay.c
index 8eb85bc..feca80d 100644 (file)
--- a/gst/rtp/gstrtph263pay.c
+++ b/gst/rtp/gstrtph263pay.c
@@ -890,9 +890,8 @@ gst_rtp_h263_pay_move_window_right (GstRtpH263PayContext * context, guint n,
     } else {
       if (n > rest_bits) {
         context->window =
-            (context->
-            window << rest_bits) | (*context->win_end & (((guint) pow (2.0,
-                        (double) rest_bits)) - 1));
+            (context->window << rest_bits) | (*context->
+            win_end & (((guint) pow (2.0, (double) rest_bits)) - 1));
         n -= rest_bits;
         rest_bits = 0;
       } else {
@@ -1654,8 +1653,8 @@ gst_rtp_h263_pay_flush (GstRtpH263Pay * rtph263pay)
 
     gst_rtp_h263_pay_boundry_init (&bound, NULL, rtph263pay->data - 1, 0, 0);
     context->gobs =
-        (GstRtpH263PayGob **) g_malloc0 (format_props[context->
-            piclayer->ptype_srcformat][0] * sizeof (GstRtpH263PayGob *));
+        (GstRtpH263PayGob **) g_malloc0 (format_props[context->piclayer->
+            ptype_srcformat][0] * sizeof (GstRtpH263PayGob *));
 
 
     for (i = 0; i < format_props[context->piclayer->ptype_srcformat][0]; i++) {

diff --git a/gst/rtp/gstrtph263pay.h b/gst/rtp/gstrtph263pay.h
index 05fb733..3dbe534 100644 (file)
--- a/gst/rtp/gstrtph263pay.h
+++ b/gst/rtp/gstrtph263pay.h
@@ -1,5 +1,5 @@
 /* GStreamer
- * Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
+ * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public

diff --git a/gst/rtp/gstrtpmparobustdepay.c b/gst/rtp/gstrtpmparobustdepay.c
index 0f3c049..62f6c9e 100644 (file)
--- a/gst/rtp/gstrtpmparobustdepay.c
+++ b/gst/rtp/gstrtpmparobustdepay.c
@@ -69,8 +69,9 @@ typedef struct _GstADUFrame
   GstBuffer *buffer;
 } GstADUFrame;
 
-GST_BOILERPLATE (GstRtpMPARobustDepay, gst_rtp_mpa_robust_depay,
-    GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD);
+#define gst_rtp_mpa_robust_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpMPARobustDepay, gst_rtp_mpa_robust_depay,
+    GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static GstStateChangeReturn gst_rtp_mpa_robust_change_state (GstElement *
     element, GstStateChange transition);
@@ -81,22 +82,6 @@ static GstBuffer *gst_rtp_mpa_robust_depay_process (GstBaseRTPDepayload *
     depayload, GstBuffer * buf);
 
 static void
-gst_rtp_mpa_robust_depay_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_mpa_robust_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_mpa_robust_depay_sink_template));
-
-  gst_element_class_set_details_simple (element_class,
-      "RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP",
-      "Extracts MPEG audio from RTP packets (RFC 5219)",
-      "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
-}
-
-static void
 gst_rtp_mpa_robust_depay_finalize (GObject * object)
 {
   GstRtpMPARobustDepay *rtpmpadepay;
@@ -109,7 +94,6 @@ gst_rtp_mpa_robust_depay_finalize (GObject * object)
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
-
 static void
 gst_rtp_mpa_robust_depay_class_init (GstRtpMPARobustDepayClass * klass)
 {
@@ -117,6 +101,9 @@ gst_rtp_mpa_robust_depay_class_init (GstRtpMPARobustDepayClass * klass)
   GstElementClass *gstelement_class;
   GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
 
+  GST_DEBUG_CATEGORY_INIT (rtpmparobustdepay_debug, "rtpmparobustdepay", 0,
+      "Robust MPEG Audio RTP Depayloader");
+
   gobject_class = (GObjectClass *) klass;
   gstelement_class = (GstElementClass *) klass;
   gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
@@ -126,16 +113,22 @@ gst_rtp_mpa_robust_depay_class_init (GstRtpMPARobustDepayClass * klass)
   gstelement_class->change_state =
       GST_DEBUG_FUNCPTR (gst_rtp_mpa_robust_change_state);
 
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_mpa_robust_depay_src_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_mpa_robust_depay_sink_template));
+
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP",
+      "Extracts MPEG audio from RTP packets (RFC 5219)",
+      "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
+
   gstbasertpdepayload_class->set_caps = gst_rtp_mpa_robust_depay_setcaps;
   gstbasertpdepayload_class->process = gst_rtp_mpa_robust_depay_process;
-
-  GST_DEBUG_CATEGORY_INIT (rtpmparobustdepay_debug, "rtpmparobustdepay", 0,
-      "Robust MPEG Audio RTP Depayloader");
 }
 
 static void
-gst_rtp_mpa_robust_depay_init (GstRtpMPARobustDepay * rtpmpadepay,
-    GstRtpMPARobustDepayClass * klass)
+gst_rtp_mpa_robust_depay_init (GstRtpMPARobustDepay * rtpmpadepay)
 {
   rtpmpadepay->adapter = gst_adapter_new ();
   rtpmpadepay->adu_frames = g_queue_new ();
@@ -280,6 +273,8 @@ gst_rtp_mpa_robust_depay_generate_dummy_frame (GstRtpMPARobustDepay *
     rtpmpadepay, GstADUFrame * frame)
 {
   GstADUFrame *dummy;
+  guint8 *data;
+  gsize size;
 
   dummy = g_slice_dup (GstADUFrame, frame);
 
@@ -292,8 +287,12 @@ gst_rtp_mpa_robust_depay_generate_dummy_frame (GstRtpMPARobustDepay *
   dummy->backpointer = 0;
 
   dummy->buffer = gst_buffer_new_and_alloc (dummy->side_info + 4);
-  memset (GST_BUFFER_DATA (dummy->buffer), 0, dummy->side_info + 4);
-  GST_WRITE_UINT32_BE (GST_BUFFER_DATA (dummy->buffer), dummy->header);
+
+  data = gst_buffer_map (dummy->buffer, &size, NULL, GST_MAP_WRITE);
+  memset (data, 0, size);
+  GST_WRITE_UINT32_BE (data, dummy->header);
+  gst_buffer_unmap (dummy->buffer, data, size);
+
   GST_BUFFER_TIMESTAMP (dummy->buffer) = GST_BUFFER_TIMESTAMP (frame->buffer);
 
   return dummy;
@@ -309,15 +308,18 @@ gst_rtp_mpa_robust_depay_queue_frame (GstRtpMPARobustDepay * rtpmpadepay,
   GstADUFrame *frame = NULL;
   guint version, layer, channels, size;
   guint crc;
+  guint8 *bdata;
+  gsize bsize;
 
   g_return_val_if_fail (buf != NULL, FALSE);
 
-  if (GST_BUFFER_SIZE (buf) < 6) {
+  bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
+
+  if (bsize < 6)
     goto corrupt_frame;
-  }
 
   frame = g_slice_new0 (GstADUFrame);
-  frame->header = GST_READ_UINT32_BE (GST_BUFFER_DATA (buf));
+  frame->header = GST_READ_UINT32_BE (bdata);
 
   size = mp3_type_frame_length_from_header (GST_ELEMENT_CAST (rtpmpadepay),
       frame->header, &version, &layer, &channels, NULL, NULL, NULL, &crc);
@@ -339,7 +341,7 @@ gst_rtp_mpa_robust_depay_queue_frame (GstRtpMPARobustDepay * rtpmpadepay,
 
   /* backpointer */
   if (layer == 3) {
-    frame->backpointer = GST_READ_UINT16_BE (GST_BUFFER_DATA (buf) + 4);
+    frame->backpointer = GST_READ_UINT16_BE (bdata + 4);
     frame->backpointer >>= 7;
     GST_LOG_OBJECT (rtpmpadepay, "backpointer: %d", frame->backpointer);
   }
@@ -351,14 +353,16 @@ gst_rtp_mpa_robust_depay_queue_frame (GstRtpMPARobustDepay * rtpmpadepay,
   frame->data_size = frame->size - 4 - frame->side_info;
 
   /* some size validation checks */
-  if (4 + frame->side_info > GST_BUFFER_SIZE (buf))
+  if (4 + frame->side_info > bsize)
     goto corrupt_frame;
 
   /* ADU data would then extend past MP3 frame,
    * even using past byte reservoir */
-  if (-frame->backpointer + (gint) (GST_BUFFER_SIZE (buf)) > frame->size)
+  if (-frame->backpointer + (gint) (bsize) > frame->size)
     goto corrupt_frame;
 
+  gst_buffer_unmap (buf, bdata, bsize);
+
   /* ok, take buffer and queue */
   frame->buffer = buf;
   g_queue_push_tail (rtpmpadepay->adu_frames, frame);
@@ -369,6 +373,7 @@ gst_rtp_mpa_robust_depay_queue_frame (GstRtpMPARobustDepay * rtpmpadepay,
 corrupt_frame:
   {
     GST_DEBUG_OBJECT (rtpmpadepay, "frame is corrupt");
+    gst_buffer_unmap (buf, bdata, bsize);
     gst_buffer_unref (buf);
     if (frame)
       g_slice_free (GstADUFrame, frame);
@@ -409,10 +414,13 @@ gst_rtp_mpa_robust_depay_deinterleave (GstRtpMPARobustDepay * rtpmpadepay,
 {
   gboolean ret = FALSE;
   guint8 *data;
+  gsize size;
   guint val, iindex, icc;
 
-  data = GST_BUFFER_DATA (buf);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
   val = GST_READ_UINT16_BE (data) >> 5;
+  gst_buffer_unmap (buf, data, size);
+
   iindex = val >> 3;
   icc = val & 0x7;
 
@@ -468,6 +476,8 @@ gst_rtp_mpa_robust_depay_push_mp3_frames (GstRtpMPARobustDepay * rtpmpadepay)
   GstFlowReturn ret = GST_FLOW_OK;
 
   while (1) {
+    guint8 *data;
+    gsize size;
 
     if (G_UNLIKELY (!rtpmpadepay->cur_adu_frame)) {
       rtpmpadepay->cur_adu_frame = rtpmpadepay->adu_frames->head;
@@ -495,7 +505,7 @@ gst_rtp_mpa_robust_depay_push_mp3_frames (GstRtpMPARobustDepay * rtpmpadepay)
       continue;
     }
 
-    if (rtpmpadepay->offset == GST_BUFFER_SIZE (frame->buffer)) {
+    if (rtpmpadepay->offset == gst_buffer_get_size (frame->buffer)) {
       if (g_list_next (rtpmpadepay->cur_adu_frame)) {
         GST_LOG_OBJECT (rtpmpadepay,
             "moving to next ADU frame, size %d, side_info %d",
@@ -523,8 +533,10 @@ gst_rtp_mpa_robust_depay_push_mp3_frames (GstRtpMPARobustDepay * rtpmpadepay)
       gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, 0);
       /* bytewriter corresponds to head frame,
        * i.e. the header and the side info must match */
+      data = gst_buffer_map (head->buffer, &size, NULL, GST_MAP_READ);
       gst_byte_writer_put_data (rtpmpadepay->mp3_frame,
-          GST_BUFFER_DATA (head->buffer), 4 + head->side_info);
+          data, 4 + head->side_info);
+      gst_buffer_unmap (head->buffer, data, size);
     }
 
     buf = frame->buffer;
@@ -534,15 +546,17 @@ gst_rtp_mpa_robust_depay_push_mp3_frames (GstRtpMPARobustDepay * rtpmpadepay)
         rtpmpadepay->size);
 
     if (rtpmpadepay->offset) {
+      data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
       /* no need to position, simply append */
-      g_assert (GST_BUFFER_SIZE (buf) > rtpmpadepay->offset);
-      av = MIN (av, GST_BUFFER_SIZE (buf) - rtpmpadepay->offset);
+      g_assert (size > rtpmpadepay->offset);
+      av = MIN (av, size - rtpmpadepay->offset);
       GST_LOG_OBJECT (rtpmpadepay,
           "appending %d bytes from ADU frame at offset %d", av,
           rtpmpadepay->offset);
       gst_byte_writer_put_data (rtpmpadepay->mp3_frame,
-          GST_BUFFER_DATA (buf) + rtpmpadepay->offset, av);
+          data + rtpmpadepay->offset, av);
       rtpmpadepay->offset += av;
+      gst_buffer_unmap (buf, data, size);
     } else {
       gint pos, tpos;
 
@@ -580,12 +594,14 @@ gst_rtp_mpa_robust_depay_push_mp3_frames (GstRtpMPARobustDepay * rtpmpadepay)
         gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, pos + av);
       } else {
         /* position and append */
+        data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
         GST_LOG_OBJECT (rtpmpadepay, "adding to current MP3 frame");
         gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, tpos);
-        av = MIN (av, GST_BUFFER_SIZE (buf) - 4 - frame->side_info);
+        av = MIN (av, size - 4 - frame->side_info);
         gst_byte_writer_put_data (rtpmpadepay->mp3_frame,
-            GST_BUFFER_DATA (buf) + 4 + frame->side_info, av);
+            data + 4 + frame->side_info, av);
         rtpmpadepay->offset += av + 4 + frame->side_info;
+        gst_buffer_unmap (buf, data, size);
       }
     }
 
@@ -634,16 +650,19 @@ gst_rtp_mpa_robust_depay_process (GstBaseRTPDepayload * depayload,
   gboolean cont, dtype;
   guint av, size;
   GstClockTime timestamp;
+  GstRTPBuffer rtp = { NULL };
 
   rtpmpadepay = GST_RTP_MPA_ROBUST_DEPAY (depayload);
 
-  payload_len = gst_rtp_buffer_get_payload_len (buf);
   timestamp = GST_BUFFER_TIMESTAMP (buf);
 
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+
+  payload_len = gst_rtp_buffer_get_payload_len (&rtp);
   if (payload_len <= 1)
     goto short_read;
 
-  payload = gst_rtp_buffer_get_payload (buf);
+  payload = gst_rtp_buffer_get_payload (&rtp);
   offset = 0;
   GST_LOG_OBJECT (rtpmpadepay, "payload_len: %d", payload_len);
 
@@ -683,7 +702,7 @@ gst_rtp_mpa_robust_depay_process (GstBaseRTPDepayload * depayload,
     GST_LOG_OBJECT (rtpmpadepay, "offset %d has cont: %d, dtype: %d, size: %d",
         offset, cont, dtype, size);
 
-    buf = gst_rtp_buffer_get_payload_subbuffer (buf, offset,
+    buf = gst_rtp_buffer_get_payload_subbuffer (&rtp, offset,
         MIN (size, payload_len));
 
     if (cont) {
@@ -693,7 +712,7 @@ gst_rtp_mpa_robust_depay_process (GstBaseRTPDepayload * depayload,
             "discarding continuation fragment without prior fragment");
         gst_buffer_unref (buf);
       } else {
-        av += GST_BUFFER_SIZE (buf);
+        av += gst_buffer_get_size (buf);
         gst_adapter_push (rtpmpadepay->adapter, buf);
         if (av == size) {
           timestamp = gst_adapter_prev_timestamp (rtpmpadepay->adapter, NULL);
@@ -727,6 +746,7 @@ gst_rtp_mpa_robust_depay_process (GstBaseRTPDepayload * depayload,
     /* timestamp applies to first payload, no idea for subsequent ones */
     timestamp = GST_CLOCK_TIME_NONE;
   }
+  gst_rtp_buffer_unmap (&rtp);
 
   return NULL;
 
@@ -735,6 +755,7 @@ short_read:
   {
     GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
         (NULL), ("Packet contains invalid data"));
+    gst_rtp_buffer_unmap (&rtp);
     return NULL;
   }
 }

diff --git a/gst/rtp/gstrtpmpvdepay.c b/gst/rtp/gstrtpmpvdepay.c
index f978604..d6e856b 100644 (file)
--- a/gst/rtp/gstrtpmpvdepay.c
+++ b/gst/rtp/gstrtpmpvdepay.c
@@ -53,8 +53,7 @@ static GstStaticPadTemplate gst_rtp_mpv_depay_sink_template =
         "clock-rate = (int) 90000")
     );
 
-GST_BOILERPLATE (GstRtpMPVDepay, gst_rtp_mpv_depay, GstBaseRTPDepayload,
-    GST_TYPE_BASE_RTP_DEPAYLOAD);
+G_DEFINE_TYPE (GstRtpMPVDepay, gst_rtp_mpv_depay, GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static gboolean gst_rtp_mpv_depay_setcaps (GstBaseRTPDepayload * depayload,
     GstCaps * caps);
@@ -62,27 +61,23 @@ static GstBuffer *gst_rtp_mpv_depay_process (GstBaseRTPDepayload * depayload,
     GstBuffer * buf);
 
 static void
-gst_rtp_mpv_depay_base_init (gpointer klass)
+gst_rtp_mpv_depay_class_init (GstRtpMPVDepayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_mpv_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_mpv_depay_sink_template));
 
-  gst_element_class_set_details_simple (element_class,
+  gst_element_class_set_details_simple (gstelement_class,
       "RTP MPEG video depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts MPEG video from RTP packets (RFC 2250)",
       "Wim Taymans <wim.taymans@gmail.com>");
-}
-
-static void
-gst_rtp_mpv_depay_class_init (GstRtpMPVDepayClass * klass)
-{
-  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
-  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
   gstbasertpdepayload_class->set_caps = gst_rtp_mpv_depay_setcaps;
   gstbasertpdepayload_class->process = gst_rtp_mpv_depay_process;
@@ -92,10 +87,8 @@ gst_rtp_mpv_depay_class_init (GstRtpMPVDepayClass * klass)
 }
 
 static void
-gst_rtp_mpv_depay_init (GstRtpMPVDepay * rtpmpvdepay,
-    GstRtpMPVDepayClass * klass)
+gst_rtp_mpv_depay_init (GstRtpMPVDepay * rtpmpvdepay)
 {
-  /* needed because of GST_BOILERPLATE */
 }
 
 static gboolean
@@ -126,16 +119,19 @@ gst_rtp_mpv_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
 {
   GstRtpMPVDepay *rtpmpvdepay;
   GstBuffer *outbuf;
+  GstRTPBuffer rtp = { NULL };
 
   rtpmpvdepay = GST_RTP_MPV_DEPAY (depayload);
 
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+
   {
     gint payload_len, payload_header;
     guint8 *payload;
     guint8 T;
 
-    payload_len = gst_rtp_buffer_get_payload_len (buf);
-    payload = gst_rtp_buffer_get_payload (buf);
+    payload_len = gst_rtp_buffer_get_payload_len (&rtp);
+    payload = gst_rtp_buffer_get_payload (&rtp);
     payload_header = 0;
 
     if (payload_len <= 4)
@@ -174,11 +170,11 @@ gst_rtp_mpv_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
       payload += 4;
     }
 
-    outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, payload_header, -1);
+    outbuf = gst_rtp_buffer_get_payload_subbuffer (&rtp, payload_header, -1);
 
     GST_DEBUG_OBJECT (rtpmpvdepay,
         "gst_rtp_mpv_depay_chain: pushing buffer of size %d",
-        GST_BUFFER_SIZE (outbuf));
+        gst_buffer_get_size (outbuf));
 
     return outbuf;
   }

diff --git a/gst/rtp/gstrtpmpvpay.c b/gst/rtp/gstrtpmpvpay.c
index a96afbf..d66968c 100644 (file)
--- a/gst/rtp/gstrtpmpvpay.c
+++ b/gst/rtp/gstrtpmpvpay.c
@@ -60,23 +60,8 @@ static GstFlowReturn gst_rtp_mpv_pay_handle_buffer (GstBaseRTPPayload *
     payload, GstBuffer * buffer);
 static gboolean gst_rtp_mpv_pay_handle_event (GstPad * pad, GstEvent * event);
 
-GST_BOILERPLATE (GstRTPMPVPay, gst_rtp_mpv_pay, GstBaseRTPPayload,
-    GST_TYPE_BASE_RTP_PAYLOAD);
-
-static void
-gst_rtp_mpv_pay_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_mpv_pay_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_mpv_pay_src_template));
-  gst_element_class_set_details_simple (element_class,
-      "RTP MPEG2 ES video payloader", "Codec/Payloader/Network/RTP",
-      "Payload-encodes MPEG2 ES into RTP packets (RFC 2250)",
-      "Thijs Vermeir <thijsvermeir@gmail.com>");
-}
+#define gst_rtp_mpv_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRTPMPVPay, gst_rtp_mpv_pay, GST_TYPE_BASE_RTP_PAYLOAD);
 
 static void
 gst_rtp_mpv_pay_class_init (GstRTPMPVPayClass * klass)
@@ -93,6 +78,16 @@ gst_rtp_mpv_pay_class_init (GstRTPMPVPayClass * klass)
 
   gstelement_class->change_state = gst_rtp_mpv_pay_change_state;
 
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_mpv_pay_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_mpv_pay_src_template));
+
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP MPEG2 ES video payloader", "Codec/Payloader/Network/RTP",
+      "Payload-encodes MPEG2 ES into RTP packets (RFC 2250)",
+      "Thijs Vermeir <thijsvermeir@gmail.com>");
+
   gstbasertppayload_class->set_caps = gst_rtp_mpv_pay_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtp_mpv_pay_handle_buffer;
   gstbasertppayload_class->handle_event = gst_rtp_mpv_pay_handle_event;
@@ -102,7 +97,7 @@ gst_rtp_mpv_pay_class_init (GstRTPMPVPayClass * klass)
 }
 
 static void
-gst_rtp_mpv_pay_init (GstRTPMPVPay * rtpmpvpay, GstRTPMPVPayClass * klass)
+gst_rtp_mpv_pay_init (GstRTPMPVPay * rtpmpvpay)
 {
   GST_BASE_RTP_PAYLOAD (rtpmpvpay)->clock_rate = 90000;
   GST_BASE_RTP_PAYLOAD_PT (rtpmpvpay) = GST_RTP_PAYLOAD_MPV;
@@ -181,6 +176,7 @@ gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
     guint towrite;
     guint packet_len;
     guint payload_len;
+    GstRTPBuffer rtp = { NULL };
 
     packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
 
@@ -190,7 +186,9 @@ gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
 
     outbuf = gst_rtp_buffer_new_allocate (payload_len, 4, 0);
 
-    payload = gst_rtp_buffer_get_payload (outbuf);
+    gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
+
+    payload = gst_rtp_buffer_get_payload (&rtp);
     /* enable MPEG Video-specific header
      *
      *  0                   1                   2                   3
@@ -211,7 +209,8 @@ gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
 
     avail -= payload_len;
 
-    gst_rtp_buffer_set_marker (outbuf, avail == 0);
+    gst_rtp_buffer_set_marker (&rtp, avail == 0);
+    gst_rtp_buffer_unmap (&rtp);
 
     GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
 

diff --git a/gst/rtp/gstrtppcmadepay.c b/gst/rtp/gstrtppcmadepay.c
index 35fc0a9..27d463f 100644 (file)
--- a/gst/rtp/gstrtppcmadepay.c
+++ b/gst/rtp/gstrtppcmadepay.c
@@ -65,38 +65,35 @@ static GstBuffer *gst_rtp_pcma_depay_process (GstBaseRTPDepayload * depayload,
 static gboolean gst_rtp_pcma_depay_setcaps (GstBaseRTPDepayload * depayload,
     GstCaps * caps);
 
-GST_BOILERPLATE (GstRtpPcmaDepay, gst_rtp_pcma_depay, GstBaseRTPDepayload,
+#define gst_rtp_pcma_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpPcmaDepay, gst_rtp_pcma_depay,
     GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static void
-gst_rtp_pcma_depay_base_init (gpointer klass)
+gst_rtp_pcma_depay_class_init (GstRtpPcmaDepayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcma_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcma_depay_sink_template));
-  gst_element_class_set_details_simple (element_class, "RTP PCMA depayloader",
-      "Codec/Depayloader/Network/RTP",
+
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP PCMA depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts PCMA audio from RTP packets",
       "Edgard Lima <edgard.lima@indt.org.br>, Zeeshan Ali <zeenix@gmail.com>");
-}
-
-static void
-gst_rtp_pcma_depay_class_init (GstRtpPcmaDepayClass * klass)
-{
-  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
-  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
   gstbasertpdepayload_class->process = gst_rtp_pcma_depay_process;
   gstbasertpdepayload_class->set_caps = gst_rtp_pcma_depay_setcaps;
 }
 
 static void
-gst_rtp_pcma_depay_init (GstRtpPcmaDepay * rtppcmadepay,
-    GstRtpPcmaDepayClass * klass)
+gst_rtp_pcma_depay_init (GstRtpPcmaDepay * rtppcmadepay)
 {
   GstBaseRTPDepayload *depayload;
 
@@ -133,15 +130,19 @@ gst_rtp_pcma_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstBuffer *outbuf = NULL;
   gboolean marker;
   guint len;
+  GstRTPBuffer rtp = { NULL };
+
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
 
-  marker = gst_rtp_buffer_get_marker (buf);
+  marker = gst_rtp_buffer_get_marker (&rtp);
 
   GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
-      GST_BUFFER_SIZE (buf), marker,
-      gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
+      gst_buffer_get_size (buf), marker,
+      gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
 
-  len = gst_rtp_buffer_get_payload_len (buf);
-  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+  len = gst_rtp_buffer_get_payload_len (&rtp);
+  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+  gst_rtp_buffer_unmap (&rtp);
 
   GST_BUFFER_DURATION (outbuf) =
       gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
@@ -151,6 +152,7 @@ gst_rtp_pcma_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
     GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
   }
 
+
   return outbuf;
 }
 

diff --git a/gst/rtp/gstrtppcmapay.c b/gst/rtp/gstrtppcmapay.c
index d897ce1..6b53dec 100644 (file)
--- a/gst/rtp/gstrtppcmapay.c
+++ b/gst/rtp/gstrtppcmapay.c
@@ -53,36 +53,34 @@ static GstStaticPadTemplate gst_rtp_pcma_pay_src_template =
 static gboolean gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload,
     GstCaps * caps);
 
-GST_BOILERPLATE (GstRtpPcmaPay, gst_rtp_pcma_pay, GstBaseRTPAudioPayload,
+#define gst_rtp_pcma_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpPcmaPay, gst_rtp_pcma_pay,
     GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
 
 static void
-gst_rtp_pcma_pay_base_init (gpointer klass)
+gst_rtp_pcma_pay_class_init (GstRtpPcmaPayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcma_pay_sink_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcma_pay_src_template));
-  gst_element_class_set_details_simple (element_class, "RTP PCMA payloader",
+
+  gst_element_class_set_details_simple (gstelement_class, "RTP PCMA payloader",
       "Codec/Payloader/Network/RTP",
       "Payload-encodes PCMA audio into a RTP packet",
       "Edgard Lima <edgard.lima@indt.org.br>");
-}
-
-static void
-gst_rtp_pcma_pay_class_init (GstRtpPcmaPayClass * klass)
-{
-  GstBaseRTPPayloadClass *gstbasertppayload_class;
-
-  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   gstbasertppayload_class->set_caps = gst_rtp_pcma_pay_setcaps;
 }
 
 static void
-gst_rtp_pcma_pay_init (GstRtpPcmaPay * rtppcmapay, GstRtpPcmaPayClass * klass)
+gst_rtp_pcma_pay_init (GstRtpPcmaPay * rtppcmapay)
 {
   GstBaseRTPAudioPayload *basertpaudiopayload;
 

diff --git a/gst/rtp/gstrtppcmudepay.c b/gst/rtp/gstrtppcmudepay.c
index fd1f1e5..0a208cd 100644 (file)
--- a/gst/rtp/gstrtppcmudepay.c
+++ b/gst/rtp/gstrtppcmudepay.c
@@ -65,38 +65,35 @@ static GstBuffer *gst_rtp_pcmu_depay_process (GstBaseRTPDepayload * depayload,
 static gboolean gst_rtp_pcmu_depay_setcaps (GstBaseRTPDepayload * depayload,
     GstCaps * caps);
 
-GST_BOILERPLATE (GstRtpPcmuDepay, gst_rtp_pcmu_depay, GstBaseRTPDepayload,
+#define gst_rtp_pcmu_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpPcmuDepay, gst_rtp_pcmu_depay,
     GST_TYPE_BASE_RTP_DEPAYLOAD);
 
 static void
-gst_rtp_pcmu_depay_base_init (gpointer klass)
+gst_rtp_pcmu_depay_class_init (GstRtpPcmuDepayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcmu_depay_src_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcmu_depay_sink_template));
-  gst_element_class_set_details_simple (element_class, "RTP PCMU depayloader",
-      "Codec/Depayloader/Network/RTP",
+
+  gst_element_class_set_details_simple (gstelement_class,
+      "RTP PCMU depayloader", "Codec/Depayloader/Network/RTP",
       "Extracts PCMU audio from RTP packets",
       "Edgard Lima <edgard.lima@indt.org.br>, Zeeshan Ali <zeenix@gmail.com>");
-}
-
-static void
-gst_rtp_pcmu_depay_class_init (GstRtpPcmuDepayClass * klass)
-{
-  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
-  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
 
   gstbasertpdepayload_class->process = gst_rtp_pcmu_depay_process;
   gstbasertpdepayload_class->set_caps = gst_rtp_pcmu_depay_setcaps;
 }
 
 static void
-gst_rtp_pcmu_depay_init (GstRtpPcmuDepay * rtppcmudepay,
-    GstRtpPcmuDepayClass * klass)
+gst_rtp_pcmu_depay_init (GstRtpPcmuDepay * rtppcmudepay)
 {
   GstBaseRTPDepayload *depayload;
 
@@ -133,15 +130,19 @@ gst_rtp_pcmu_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
   GstBuffer *outbuf = NULL;
   guint len;
   gboolean marker;
+  GstRTPBuffer rtp = { NULL };
+
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
 
-  marker = gst_rtp_buffer_get_marker (buf);
+  marker = gst_rtp_buffer_get_marker (&rtp);
 
   GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
-      GST_BUFFER_SIZE (buf), marker,
-      gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
+      gst_buffer_get_size (buf), marker,
+      gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
 
-  len = gst_rtp_buffer_get_payload_len (buf);
-  outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+  len = gst_rtp_buffer_get_payload_len (&rtp);
+  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+  gst_rtp_buffer_unmap (&rtp);
 
   GST_BUFFER_DURATION (outbuf) =
       gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);

diff --git a/gst/rtp/gstrtppcmupay.c b/gst/rtp/gstrtppcmupay.c
index e7a0995..6b40a20 100644 (file)
--- a/gst/rtp/gstrtppcmupay.c
+++ b/gst/rtp/gstrtppcmupay.c
@@ -53,36 +53,34 @@ static GstStaticPadTemplate gst_rtp_pcmu_pay_src_template =
 static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
     GstCaps * caps);
 
-GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload,
+#define gst_rtp_pcmu_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpPcmuPay, gst_rtp_pcmu_pay,
     GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
 
 static void
-gst_rtp_pcmu_pay_base_init (gpointer klass)
+gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcmu_pay_sink_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_pcmu_pay_src_template));
-  gst_element_class_set_details_simple (element_class, "RTP PCMU payloader",
+
+  gst_element_class_set_details_simple (gstelement_class, "RTP PCMU payloader",
       "Codec/Payloader/Network/RTP",
       "Payload-encodes PCMU audio into a RTP packet",
       "Edgard Lima <edgard.lima@indt.org.br>");
-}
-
-static void
-gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
-{
-  GstBaseRTPPayloadClass *gstbasertppayload_class;
-
-  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
 
   gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
 }
 
 static void
-gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
+gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay)
 {
   GstBaseRTPAudioPayload *basertpaudiopayload;