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, 149 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpspeexpay.c b/gst/rtp/gstrtpspeexpay.c
new file mode 100644
index 00000000..97e3bf33
--- /dev/null
+++ b/gst/rtp/gstrtpspeexpay.c
@@ -0,0 +1,149 @@
+/* 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 <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpspeexenc.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtpspeexenc_details = {
+  "RTP packet parser",
+  "Codec/Encoder/Network",
+  "Encodes Speex audio into a RTP packet",
+  "Edgard Lima <edgard.lima@indt.org.br>"
+};
+
+static GstStaticPadTemplate gst_rtpspeexenc_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-speex")
+    );
+
+static GstStaticPadTemplate gst_rtpspeexenc_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) 110, "       /* guaranties compatibility with Linphone
+                                                                                                           Could be [96,127] See page 34 at http://www.ietf.org/rfc/rfc3551.txt */
+        "clock-rate =  (int) [6000, 48000], "
+        "encoding-name = (string) \"speex\", "
+        "encoding-params = (string) \"1\"")
+    );
+
+static gboolean gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+static GstFlowReturn gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * payload,
+    GstBuffer * buffer);
+
+GST_BOILERPLATE (GstRtpSPEEXEnc, gst_rtpspeexenc, GstBaseRTPPayload,
+    GST_TYPE_BASE_RTP_PAYLOAD);
+
+static void
+gst_rtpspeexenc_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_rtpspeexenc_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpspeexenc_src_template));
+  gst_element_class_set_details (element_class, &gst_rtpspeexenc_details);
+}
+
+static void
+gst_rtpspeexenc_class_init (GstRtpSPEEXEncClass * 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_rtpspeexenc_setcaps;
+  gstbasertppayload_class->handle_buffer = gst_rtpspeexenc_handle_buffer;
+}
+
+static void
+gst_rtpspeexenc_init (GstRtpSPEEXEnc * rtpspeexenc, GstRtpSPEEXEncClass * klass)
+{
+  GST_BASE_RTP_PAYLOAD (rtpspeexenc)->clock_rate = 8000;
+  GST_BASE_RTP_PAYLOAD_PT (rtpspeexenc) = 110;  /* Create String */
+}
+
+static gboolean
+gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+  gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
+  gst_basertppayload_set_outcaps (payload, NULL);
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstRtpSPEEXEnc *rtpspeexenc;
+  guint size, payload_len;
+  GstBuffer *outbuf;
+  guint8 *payload, *data;
+  GstClockTime timestamp;
+  GstFlowReturn ret;
+
+  rtpspeexenc = GST_RTP_SPEEX_ENC (basepayload);
+
+  size = GST_BUFFER_SIZE (buffer);
+  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+  /* FIXME, only one SPEEX frame per RTP packet for now */
+  payload_len = size;
+
+  outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
+  /* FIXME, assert for now */
+  g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexenc));
+
+  /* copy timestamp */
+  GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+  /* get payload */
+  payload = gst_rtpbuffer_get_payload (outbuf);
+
+  data = GST_BUFFER_DATA (buffer);
+
+  /* copy data in payload */
+  memcpy (&payload[0], data, size);
+
+  gst_buffer_unref (buffer);
+
+  ret = gst_basertppayload_push (basepayload, outbuf);
+
+  return ret;
+}
+
+gboolean
+gst_rtpspeexenc_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpspeexenc",
+      GST_RANK_NONE, GST_TYPE_RTP_SPEEX_ENC);
+}