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/* 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 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpspeexdepay.h"
#include "gstrtputils.h"
/* RtpSPEEXDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
static GstStaticPadTemplate gst_rtp_speex_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"clock-rate = (int) [6000, 48000], "
"encoding-name = (string) \"SPEEX\"")
/* "encoding-params = (string) \"1\"" */
);
static GstStaticPadTemplate gst_rtp_speex_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstBuffer *gst_rtp_speex_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static gboolean gst_rtp_speex_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
G_DEFINE_TYPE (GstRtpSPEEXDepay, gst_rtp_speex_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_speex_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_speex_depay_setcaps;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_speex_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_speex_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP Speex depayloader", "Codec/Depayloader/Network/RTP",
"Extracts Speex audio from RTP packets",
"Edgard Lima <edgard.lima@indt.org.br>");
}
static void
gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay)
{
}
static gint
gst_rtp_speex_depay_get_mode (gint rate)
{
if (rate > 25000)
return 2;
else if (rate > 12500)
return 1;
else
return 0;
}
/* len 4 bytes LE,
* vendor string (len bytes),
* user_len 4 (0) bytes LE
*/
static const gchar gst_rtp_speex_comment[] =
"\045\0\0\0Depayloaded with GStreamer speexdepay\0\0\0\0";
static gboolean
gst_rtp_speex_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpSPEEXDepay *rtpspeexdepay;
gint clock_rate, nb_channels;
GstBuffer *buf;
GstMapInfo map;
guint8 *data;
const gchar *params;
GstCaps *srccaps;
gboolean res;
rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
goto no_clockrate;
depayload->clock_rate = clock_rate;
if (!(params = gst_structure_get_string (structure, "encoding-params")))
nb_channels = 1;
else {
nb_channels = atoi (params);
}
/* construct minimal header and comment packet for the decoder */
buf = gst_buffer_new_and_alloc (80);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
data = map.data;
memcpy (data, "Speex ", 8);
data += 8;
memcpy (data, "1.1.12", 7);
data += 20;
GST_WRITE_UINT32_LE (data, 1); /* version */
data += 4;
GST_WRITE_UINT32_LE (data, 80); /* header_size */
data += 4;
GST_WRITE_UINT32_LE (data, clock_rate); /* rate */
data += 4;
GST_WRITE_UINT32_LE (data, gst_rtp_speex_depay_get_mode (clock_rate)); /* mode */
data += 4;
GST_WRITE_UINT32_LE (data, 4); /* mode_bitstream_version */
data += 4;
GST_WRITE_UINT32_LE (data, nb_channels); /* nb_channels */
data += 4;
GST_WRITE_UINT32_LE (data, -1); /* bitrate */
data += 4;
GST_WRITE_UINT32_LE (data, 0xa0); /* frame_size */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* VBR */
data += 4;
GST_WRITE_UINT32_LE (data, 1); /* frames_per_packet */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* extra_headers */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* reserved1 */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* reserved2 */
gst_buffer_unmap (buf, &map);
srccaps = gst_caps_new_empty_simple ("audio/x-speex");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpspeexdepay), buf);
buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment));
gst_buffer_fill (buf, 0, gst_rtp_speex_comment,
sizeof (gst_rtp_speex_comment));
gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpspeexdepay), buf);
return res;
/* ERRORS */
no_clockrate:
{
GST_DEBUG_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static GstBuffer *
gst_rtp_speex_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp)
{
GstBuffer *outbuf = NULL;
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer),
gst_rtp_buffer_get_marker (rtp),
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
/* nothing special to be done */
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (outbuf) {
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
return outbuf;
}
gboolean
gst_rtp_speex_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_DEPAY);
}
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