1 files changed, 140 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpopuspay.c b/gst/rtp/gstrtpopuspay.c
new file mode 100644
index 000000000..9724e6276
--- /dev/null
+++ b/gst/rtp/gstrtpopuspay.c
@@ -0,0 +1,140 @@
+/*
+ * Opus Payloader Gst Element
+ *
+ *   @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpopuspay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
+#define GST_CAT_DEFAULT (rtpopuspay_debug)
+
+
+static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-opus")
+    );
+
+static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) 48000, "
+        "encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
+    );
+
+static gboolean gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload *
+    payload, GstBuffer * buffer);
+
+GST_BOILERPLATE (GstRtpOPUSPay, gst_rtp_opus_pay, GstBaseRTPPayload,
+    GST_TYPE_BASE_RTP_PAYLOAD);
+
+static void
+gst_rtp_opus_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_opus_pay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template));
+
+  gst_element_class_set_details_simple (element_class,
+      "RTP Opus payloader",
+      "Codec/Payloader/Network/RTP",
+      "Puts Opus audio in RTP packets",
+      "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
+}
+
+static void
+gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
+{
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
+  gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
+
+  GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
+      "Opus RTP Payloader");
+}
+
+static void
+gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay, GstRtpOPUSPayClass * klass)
+{
+}
+
+static gboolean
+gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+  gboolean res;
+  gchar *capsstr;
+
+  capsstr = gst_caps_to_string (caps);
+
+  gst_basertppayload_set_options (payload, "audio", FALSE,
+      "X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
+  res =
+      gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr,
+      NULL);
+  g_free (capsstr);
+
+  return res;
+}
+
+static GstFlowReturn
+gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstBuffer *outbuf;
+  GstClockTime timestamp;
+
+  guint size;
+  guint8 *data;
+  guint8 *payload;
+
+  size = GST_BUFFER_SIZE (buffer);
+  data = GST_BUFFER_DATA (buffer);
+  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+  outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
+  payload = gst_rtp_buffer_get_payload (outbuf);
+
+  memcpy (payload, data, size);
+
+  gst_rtp_buffer_set_marker (outbuf, FALSE);
+  GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+
+  return gst_basertppayload_push (basepayload, outbuf);
+}