gstreamer/gst/rtp/gstrtpspeexenc.c

/* 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);
}