gst/rtp/: Ported mulaw and alaw payloaders to use new base class

Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpilbcpay.h:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcdepay.h:
Added new iLBC payloader/depayloader. Payloader uses new audio payload base
class.

1 files changed, 161 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpilbcpay.c b/gst/rtp/gstrtpilbcpay.c
new file mode 100644
index 00000000..7dbac86c
--- /dev/null
+++ b/gst/rtp/gstrtpilbcpay.c
@@ -0,0 +1,161 @@
+/* GStreamer
+ * Copyright (C) <2006> Philippe Khalaf <burger@speedy.org> 
+ *
+ * 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 "gstrtpilbcpay.h"
+#include <gst/rtp/gstrtpbuffer.h>
+
+/* elementfactory information */
+static GstElementDetails gst_rtpilbcpay_details = {
+  "RTP Payloader for iLBC Audio",
+  "Codec/Payloader/Network",
+  "Packetize iLBC audio streams into RTP packets",
+  "Philippe Kalaf <philippe.kalaf@collabora.co.uk>"
+};
+
+GST_DEBUG_CATEGORY (rtpilbcpay_debug);
+#define GST_CAT_DEFAULT (rtpilbcpay_debug)
+
+static GstStaticPadTemplate gst_rtpilbcpay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-iLBC, " "mode = (int) {20, 30}")
+    );
+
+static GstStaticPadTemplate gst_rtpilbcpay_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) \"iLBC\", " "mode = (int) {20, 30}")
+    );
+
+static gboolean gst_rtpilbcpay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRTPILBCPay, gst_rtpilbcpay, GstBaseRTPAudioPayload,
+    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtpilbcpay_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_rtpilbcpay_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpilbcpay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtpilbcpay_details);
+}
+
+static void
+gst_rtpilbcpay_class_init (GstRTPILBCPayClass * 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_rtpilbcpay_setcaps;
+
+  GST_DEBUG_CATEGORY_INIT (rtpilbcpay_debug, "rtpilbcpay", 0,
+      "iLBC audio RTP payloader");
+}
+
+static void
+gst_rtpilbcpay_init (GstRTPILBCPay * rtpilbcpay, GstRTPILBCPayClass * klass)
+{
+  GstBaseRTPPayload *basertppayload;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertppayload = GST_BASE_RTP_PAYLOAD (rtpilbcpay);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpilbcpay);
+
+  /* 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;
+
+  rtpilbcpay->mode = -1;
+
+  /* tell basertpaudiopayload that this is a frame based codec */
+  gst_basertpaudiopayload_set_frame_based (basertpaudiopayload);
+}
+
+static gboolean
+gst_rtpilbcpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+{
+  GstRTPILBCPay *rtpilbcpay;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  gboolean ret;
+  gint mode;
+  GstStructure *structure;
+  const char *payload_name;
+
+  rtpilbcpay = GST_RTP_ILBC_PAY (basertppayload);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  gst_structure_get_int (structure, "mode", &mode);
+  if (mode != 20 && mode != 30) {
+    return FALSE;
+  }
+
+  payload_name = gst_structure_get_name (structure);
+  if (g_strcasecmp ("audio/x-iLBC", payload_name) == 0) {
+    gst_basertppayload_set_options (basertppayload, "audio", TRUE, "iLBC",
+        8000);
+    /* set options for this frame based audio codec */
+    gst_basertpaudiopayload_set_frame_options (basertpaudiopayload,
+        mode, mode == 30 ? 50 : 38);
+  } else {
+    return FALSE;
+  }
+
+  ret =
+      gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_INT, mode,
+      NULL);
+  if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1) {
+    GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! \
+          Mode cannot change while streaming", rtpilbcpay->mode, mode);
+    return FALSE;
+  }
+  rtpilbcpay->mode = mode;
+
+  return ret;
+}
+
+gboolean
+gst_rtp_ilbc_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpilbcpay",
+      GST_RANK_NONE, GST_TYPE_RTP_ILBC_PAY);
+}