mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
For more optimised RTP packet handling: means we don't need to map the input buffer again but can just re-use the mapping the base class has already done. https://bugzilla.gnome.org/show_bug.cgi?id=750235
222 lines
6.3 KiB
C
222 lines
6.3 KiB
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 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 "gstrtpspeexdepay.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_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_rtp_speex_depay_src_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&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;
|
|
|
|
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);
|
|
}
|