gst/rtp/: Added G729 pay and depayloaders. Fixes #532409.

Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_base_init),
(gst_rtp_g729_depay_class_init), (gst_rtp_g729_depay_init),
(gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process),
(gst_rtp_g729_depay_plugin_init):
* gst/rtp/gstrtpg729depay.h:
* gst/rtp/gstrtpg729pay.c: (gst_rtpg729pay_base_init),
(gst_rtpg729pay_class_init), (gst_rtpg729pay_init),
(gst_rtpg729pay_setcaps), (gst_rtp_g729_pay_plugin_init):
* gst/rtp/gstrtpg729pay.h:
Added G729 pay and depayloaders. Fixes #532409.

1 files changed, 150 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpg729pay.c b/gst/rtp/gstrtpg729pay.c
new file mode 100644
index 00000000..b2f0dd90
--- /dev/null
+++ b/gst/rtp/gstrtpg729pay.c
@@ -0,0 +1,150 @@
+/* GStreamer
+ *
+ * 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 "gstrtpg729pay.h"
+#include <gst/rtp/gstrtpbuffer.h>
+
+/* elementfactory information */
+static GstElementDetails gst_rtpg729pay_details = {
+  "RTP Payloader for G729 Audio",
+  "Codec/Payloader/Network",
+  "Packetize G729 audio streams into RTP packets",
+  "Laurent Glayal <spglegle@yahoo.fr>"
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
+#define GST_CAT_DEFAULT (rtpg729pay_debug)
+
+static GstStaticPadTemplate gst_rtpg729pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/G729, channels=(int)1, rate=(int)8000")
+    );
+
+static GstStaticPadTemplate gst_rtpg729pay_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) 8000, " "encoding-name = (string) \"G729\";"
+        "application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
+        "clock-rate = (int) 8000")
+    );
+
+static gboolean gst_rtpg729pay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRtpG729Pay, gst_rtpg729pay, GstBaseRTPAudioPayload,
+    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtpg729pay_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_rtpg729pay_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpg729pay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtpg729pay_details);
+}
+
+static void
+gst_rtpg729pay_class_init (GstRtpG729PayClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+
+  gstbasertppayload_class->set_caps = gst_rtpg729pay_setcaps;
+
+  GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
+      "G729 audio RTP payloader");
+}
+
+static void
+gst_rtpg729pay_init (GstRtpG729Pay * rtpg729pay, GstRtpG729PayClass * klass)
+{
+  GstBaseRTPPayload *basertppayload;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertppayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg729pay);
+
+  /* we don't set the payload type, it should be set by the application using
+   * the pt property or the default 96 will be used */
+  basertppayload->clock_rate = 8000;
+
+  /* tell basertpaudiopayload that this is a frame based codec */
+  gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
+  gst_basertppayload_set_options (basertppayload, "audio", FALSE, "G729", 8000);
+  gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 10, 10);
+}
+
+static gboolean
+gst_rtpg729pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+{
+  GstRtpG729Pay *rtpg729pay;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  gboolean ret;
+  GstStructure *structure;
+  const char *payload_name;
+
+  rtpg729pay = GST_RTP_G729_PAY (basertppayload);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  payload_name = gst_structure_get_name (structure);
+  if (g_strcasecmp ("audio/G729", payload_name) != 0)
+    goto wrong_name;
+
+  ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
+
+  return ret;
+
+  /* ERRORS */
+wrong_name:
+  {
+    GST_ERROR_OBJECT (rtpg729pay, "wrong name, expected 'audio/G729', got '%s'",
+        payload_name);
+    return FALSE;
+  }
+}
+
+gboolean
+gst_rtp_g729_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpg729pay",
+      GST_RANK_NONE, GST_TYPE_RTP_G729_PAY);
+}