Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.

Original commit message from CVS:
Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.

1 files changed, 143 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpspeexdepay.c b/gst/rtp/gstrtpspeexdepay.c
new file mode 100644
index 00000000..febcef4a
--- /dev/null
+++ b/gst/rtp/gstrtpspeexdepay.c
@@ -0,0 +1,143 @@
+/* GStreamer
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ *
+ * 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 
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpspeexdec.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtp_speexdec_details = {
+  "RTP packet parser",
+  "Codec/Parser/Network",
+  "Extracts Speex audio from RTP packets",
+  "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+/* RtpSPEEXDec signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  ARG_0
+};
+
+static GstStaticPadTemplate gst_rtpspeexdec_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) [ 96, 127 ], "
+        "clock-rate =  (int) [6000, 48000], "
+        "encoding-name = (string) \"speex\", "
+        "encoding-params = (string) \"1\"")
+    );
+
+static GstStaticPadTemplate gst_rtpspeexdec_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-speex")
+    );
+
+static GstBuffer *gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload,
+    GstBuffer * buf);
+static gboolean gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRtpSPEEXDec, gst_rtpspeexdec, GstBaseRTPDepayload,
+    GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void
+gst_rtpspeexdec_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_rtpspeexdec_src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpspeexdec_sink_template));
+  gst_element_class_set_details (element_class, &gst_rtp_speexdec_details);
+}
+
+static void
+gst_rtpspeexdec_class_init (GstRtpSPEEXDecClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+  gstbasertpdepayload_class->process = gst_rtpspeexdec_process;
+  gstbasertpdepayload_class->set_caps = gst_rtpspeexdec_setcaps;
+}
+
+static void
+gst_rtpspeexdec_init (GstRtpSPEEXDec * rtpspeexdec, GstRtpSPEEXDecClass * klass)
+{
+  GST_BASE_RTP_DEPAYLOAD (rtpspeexdec)->clock_rate = 8000;
+}
+
+static gboolean
+gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+  GstCaps *srccaps;
+  gboolean ret;
+
+  srccaps = gst_static_pad_template_get_caps (&gst_rtpspeexdec_src_template);
+  ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+  gst_caps_unref (srccaps);
+  return ret;
+}
+
+static GstBuffer *
+gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+  GstBuffer *outbuf = NULL;
+  gint payload_len;
+  guint8 *payload;
+
+  GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
+      GST_BUFFER_SIZE (buf),
+      gst_rtpbuffer_get_marker (buf),
+      gst_rtpbuffer_get_timestamp (buf), gst_rtpbuffer_get_seq (buf));
+
+  payload_len = gst_rtpbuffer_get_payload_len (buf);
+  payload = gst_rtpbuffer_get_payload (buf);
+
+  outbuf = gst_buffer_new_and_alloc (payload_len);
+  memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
+  return outbuf;
+}
+
+gboolean
+gst_rtpspeexdec_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpspeexdec",
+      GST_RANK_NONE, GST_TYPE_RTP_SPEEX_DEC);
+}