Merge branch 'plugin-move-rtp-opus'

Move Opus RTP depayloader/payloader from -bad to -good.

https://bugzilla.gnome.org/show_bug.cgi?id=756282

4 files changed, 568 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpopusdepay.c b/gst/rtp/gstrtpopusdepay.c
new file mode 100644
index 000000000..8152cd57f
--- /dev/null
+++ b/gst/rtp/gstrtpopusdepay.c
@@ -0,0 +1,175 @@
+/*
+ * Opus Depayloader 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <string.h>
+#include <stdlib.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
+#include "gstrtpopusdepay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug);
+#define GST_CAT_DEFAULT (rtpopusdepay_debug)
+
+static GstStaticPadTemplate gst_rtp_opus_depay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    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) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
+    );
+
+static GstStaticPadTemplate gst_rtp_opus_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) 0")
+    );
+
+static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
+    GstBuffer * buf);
+static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
+    GstCaps * caps);
+
+G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
+    GST_TYPE_RTP_BASE_DEPAYLOAD);
+
+static void
+gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
+{
+  GstRTPBaseDepayloadClass *gstbasertpdepayload_class;
+  GstElementClass *element_class;
+
+  element_class = GST_ELEMENT_CLASS (klass);
+  gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass;
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_opus_depay_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_opus_depay_sink_template));
+  gst_element_class_set_static_metadata (element_class,
+      "RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP",
+      "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->set_caps = gst_rtp_opus_depay_setcaps;
+
+  GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
+      "Opus RTP Depayloader");
+}
+
+static void
+gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay)
+{
+
+}
+
+static gboolean
+gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
+{
+  GstCaps *srccaps;
+  GstStructure *s;
+  gboolean ret;
+  const gchar *sprop_stereo, *sprop_maxcapturerate;
+
+  srccaps =
+      gst_caps_new_simple ("audio/x-opus", "channel-mapping-family", G_TYPE_INT,
+      0, NULL);
+
+  s = gst_caps_get_structure (caps, 0);
+  if ((sprop_stereo = gst_structure_get_string (s, "sprop-stereo"))) {
+    if (strcmp (sprop_stereo, "0") == 0)
+      gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 1, NULL);
+    else if (strcmp (sprop_stereo, "1") == 0)
+      gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL);
+    else
+      GST_WARNING_OBJECT (depayload, "Unknown sprop-stereo value '%s'",
+          sprop_stereo);
+  }
+
+  if ((sprop_maxcapturerate =
+          gst_structure_get_string (s, "sprop-maxcapturerate"))) {
+    gulong rate;
+    gchar *tailptr;
+
+    rate = strtoul (sprop_maxcapturerate, &tailptr, 10);
+    if (rate > INT_MAX || *tailptr != '\0') {
+      GST_WARNING_OBJECT (depayload,
+          "Failed to parse sprop-maxcapturerate value '%s'",
+          sprop_maxcapturerate);
+    } else {
+      gst_caps_set_simple (srccaps, "rate", G_TYPE_INT, rate, NULL);
+    }
+  }
+
+  ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+  GST_DEBUG_OBJECT (depayload,
+      "set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
+  gst_caps_unref (srccaps);
+
+  depayload->clock_rate = 48000;
+
+  return ret;
+}
+
+static gboolean
+foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
+{
+  GstRTPOpusDepay *depay = user_data;
+  const GstMetaInfo *info = (*meta)->info;
+  const gchar *const *tags = gst_meta_api_type_get_tags (info->api);
+
+  if (!tags || (g_strv_length ((gchar **) tags) == 1
+          && gst_meta_api_type_has_tag (info->api,
+              g_quark_from_string (GST_META_TAG_AUDIO_STR)))) {
+    GST_DEBUG_OBJECT (depay, "keeping metadata %s", g_type_name (info->api));
+  } else {
+    GST_DEBUG_OBJECT (depay, "dropping metadata %s", g_type_name (info->api));
+    *meta = NULL;
+  }
+
+  return TRUE;
+}
+
+static GstBuffer *
+gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
+{
+  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_buffer_make_writable (outbuf);
+  /* Filter away all metas that are not sensible to copy */
+  gst_buffer_foreach_meta (outbuf, foreach_metadata, depayload);
+
+  return outbuf;
+}
diff --git a/gst/rtp/gstrtpopusdepay.h b/gst/rtp/gstrtpopusdepay.h
new file mode 100644
index 000000000..7890eb19d
--- /dev/null
+++ b/gst/rtp/gstrtpopusdepay.h
@@ -0,0 +1,57 @@
+/*
+ * Opus Depayloader Gst Element
+ *
+ *   @author: Danilo Cesar Lemes de Paula <danilo.eu@gmail.com>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_OPUS_DEPAY_H__
+#define __GST_RTP_OPUS_DEPAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtpbasedepayload.h>
+
+G_BEGIN_DECLS typedef struct _GstRTPOpusDepay GstRTPOpusDepay;
+typedef struct _GstRTPOpusDepayClass GstRTPOpusDepayClass;
+
+#define GST_TYPE_RTP_OPUS_DEPAY \
+  (gst_rtp_opus_depay_get_type())
+#define GST_RTP_OPUS_DEPAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_OPUS_DEPAY,GstRTPOpusDepay))
+#define GST_RTP_OPUS_DEPAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_OPUS_DEPAY,GstRTPOpusDepayClass))
+#define GST_IS_RTP_OPUS_DEPAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_OPUS_DEPAY))
+#define GST_IS_RTP_OPUS_DEPAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_OPUS_DEPAY))
+
+
+struct _GstRTPOpusDepay
+{
+  GstRTPBaseDepayload depayload;
+
+};
+
+struct _GstRTPOpusDepayClass
+{
+  GstRTPBaseDepayloadClass parent_class;
+};
+
+GType gst_rtp_opus_depay_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_RTP_OPUS_DEPAY_H__ */
diff --git a/gst/rtp/gstrtpopuspay.c b/gst/rtp/gstrtpopuspay.c
new file mode 100644
index 000000000..5038028b1
--- /dev/null
+++ b/gst/rtp/gstrtpopuspay.c
@@ -0,0 +1,278 @@
+/*
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.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, channels = (int) [1, 2], channel-mapping-family = (int) 0")
+    );
+
+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-params = (string) \"2\", "
+        "encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
+    );
+
+static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
+    GstCaps * caps);
+static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
+    GstPad * pad, GstCaps * filter);
+static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
+    payload, GstBuffer * buffer);
+
+G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
+
+static void
+gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
+{
+  GstRTPBasePayloadClass *gstbasertppayload_class;
+  GstElementClass *element_class;
+
+  gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
+  element_class = GST_ELEMENT_CLASS (klass);
+
+  gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
+  gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
+  gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
+
+  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_static_metadata (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>");
+
+  GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
+      "Opus RTP Payloader");
+}
+
+static void
+gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
+{
+}
+
+static gboolean
+gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
+{
+  gboolean res;
+  GstCaps *src_caps;
+  GstStructure *s;
+  char *encoding_name;
+  gint channels, rate;
+  const char *sprop_stereo = NULL;
+  char *sprop_maxcapturerate = NULL;
+
+  src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
+  if (src_caps) {
+    src_caps = gst_caps_make_writable (src_caps);
+    src_caps = gst_caps_truncate (src_caps);
+    s = gst_caps_get_structure (src_caps, 0);
+    gst_structure_fixate_field_string (s, "encoding-name", "OPUS");
+    encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name"));
+    gst_caps_unref (src_caps);
+  } else {
+    encoding_name = g_strdup ("X-GST-OPUS-DRAFT-SPITTKA-00");
+  }
+
+  s = gst_caps_get_structure (caps, 0);
+  if (gst_structure_get_int (s, "channels", &channels)) {
+    if (channels > 2) {
+      GST_ERROR_OBJECT (payload,
+          "More than 2 channels with channel-mapping-family=0 is invalid");
+      return FALSE;
+    } else if (channels == 2) {
+      sprop_stereo = "1";
+    } else {
+      sprop_stereo = "0";
+    }
+  }
+
+  if (gst_structure_get_int (s, "rate", &rate)) {
+    sprop_maxcapturerate = g_strdup_printf ("%d", rate);
+  }
+
+  gst_rtp_base_payload_set_options (payload, "audio", FALSE,
+      encoding_name, 48000);
+  g_free (encoding_name);
+
+  if (sprop_maxcapturerate && sprop_stereo) {
+    res =
+        gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
+        G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING,
+        sprop_stereo, NULL);
+  } else if (sprop_maxcapturerate) {
+    res =
+        gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
+        G_TYPE_STRING, sprop_maxcapturerate, NULL);
+  } else if (sprop_stereo) {
+    res =
+        gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo",
+        G_TYPE_STRING, sprop_stereo, NULL);
+  } else {
+    res = gst_rtp_base_payload_set_outcaps (payload, NULL);
+  }
+
+  g_free (sprop_maxcapturerate);
+
+  return res;
+}
+
+typedef struct
+{
+  GstRtpOPUSPay *pay;
+  GstBuffer *outbuf;
+} CopyMetaData;
+
+static gboolean
+foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
+{
+  CopyMetaData *data = user_data;
+  GstRtpOPUSPay *pay = data->pay;
+  GstBuffer *outbuf = data->outbuf;
+  const GstMetaInfo *info = (*meta)->info;
+  const gchar *const *tags = gst_meta_api_type_get_tags (info->api);
+
+  if (!tags || (g_strv_length ((gchar **) tags) == 1
+          && gst_meta_api_type_has_tag (info->api,
+              g_quark_from_string (GST_META_TAG_AUDIO_STR)))) {
+    GstMetaTransformCopy copy_data = { FALSE, 0, -1 };
+    GST_DEBUG_OBJECT (pay, "copy metadata %s", g_type_name (info->api));
+    /* simply copy then */
+    info->transform_func (outbuf, *meta, inbuf,
+        _gst_meta_transform_copy, &copy_data);
+  } else {
+    GST_DEBUG_OBJECT (pay, "not copying metadata %s", g_type_name (info->api));
+  }
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstBuffer *outbuf;
+  GstClockTime pts, dts, duration;
+  CopyMetaData data;
+
+  pts = GST_BUFFER_PTS (buffer);
+  dts = GST_BUFFER_DTS (buffer);
+  duration = GST_BUFFER_DURATION (buffer);
+
+  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  data.pay = GST_RTP_OPUS_PAY (basepayload);
+  data.outbuf = outbuf;
+  gst_buffer_foreach_meta (buffer, foreach_metadata, &data);
+  outbuf = gst_buffer_append (outbuf, buffer);
+
+  GST_BUFFER_PTS (outbuf) = pts;
+  GST_BUFFER_DTS (outbuf) = dts;
+  GST_BUFFER_DURATION (outbuf) = duration;
+
+  /* Push out */
+  return gst_rtp_base_payload_push (basepayload, outbuf);
+}
+
+static GstCaps *
+gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
+    GstPad * pad, GstCaps * filter)
+{
+  GstCaps *caps, *peercaps, *tcaps;
+  GstStructure *s;
+  const gchar *stereo;
+
+  if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))
+    return
+        GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
+        (payload, pad, filter);
+
+  tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
+  peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),
+      tcaps);
+  gst_caps_unref (tcaps);
+  if (!peercaps)
+    return
+        GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
+        (payload, pad, filter);
+
+  if (gst_caps_is_empty (peercaps))
+    return peercaps;
+
+  caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));
+
+  s = gst_caps_get_structure (peercaps, 0);
+  stereo = gst_structure_get_string (s, "stereo");
+  if (stereo != NULL) {
+    caps = gst_caps_make_writable (caps);
+
+    if (!strcmp (stereo, "1")) {
+      GstCaps *caps2 = gst_caps_copy (caps);
+
+      gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
+      gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);
+      caps = gst_caps_merge (caps, caps2);
+    } else if (!strcmp (stereo, "0")) {
+      GstCaps *caps2 = gst_caps_copy (caps);
+
+      gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
+      gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);
+      caps = gst_caps_merge (caps, caps2);
+    }
+  }
+  gst_caps_unref (peercaps);
+
+  if (filter) {
+    GstCaps *tmp = gst_caps_intersect_full (caps, filter,
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (caps);
+    caps = tmp;
+  }
+
+  GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);
+  return caps;
+}
diff --git a/gst/rtp/gstrtpopuspay.h b/gst/rtp/gstrtpopuspay.h
new file mode 100644
index 000000000..e21bbe3c3
--- /dev/null
+++ b/gst/rtp/gstrtpopuspay.h
@@ -0,0 +1,58 @@
+/*
+ * Opus Payloader Gst Element
+ *
+ *   @author: Danilo Cesar Lemes de Paula <danilo.eu@gmail.com>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_OPUS_PAY_H__
+#define __GST_RTP_OPUS_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtpbasepayload.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_OPUS_PAY \
+  (gst_rtp_opus_pay_get_type())
+#define GST_RTP_OPUS_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_OPUS_PAY,GstRtpOPUSPay))
+#define GST_RTP_OPUS_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_OPUS_PAY,GstRtpOPUSPayClass))
+#define GST_IS_RTP_OPUS_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_OPUS_PAY))
+#define GST_IS_RTP_OPUS_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_OPUS_PAY))
+
+typedef struct _GstRtpOPUSPay GstRtpOPUSPay;
+typedef struct _GstRtpOPUSPayClass GstRtpOPUSPayClass;
+
+struct _GstRtpOPUSPay
+{
+  GstRTPBasePayload payload;
+};
+
+struct _GstRtpOPUSPayClass
+{
+  GstRTPBasePayloadClass parent_class;
+};
+
+GType gst_rtp_opus_pay_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_OPUS_PAY_H__ */