mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-30 04:00:37 +00:00
50f88db3ad
Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index): * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send): Fix some compiler warnings. Fixes #428182.
216 lines
6.3 KiB
C
216 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., 59 Temple Place - Suite 330,
|
|
* Boston, MA 02111-1307, USA.
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include "config.h"
|
|
#endif
|
|
|
|
#include <string.h>
|
|
#include <stdlib.h>
|
|
#include <gst/rtp/gstrtpbuffer.h>
|
|
|
|
#include "gstrtpspeexdepay.h"
|
|
|
|
/* elementfactory information */
|
|
static const GstElementDetails gst_rtp_speexdepay_details =
|
|
GST_ELEMENT_DETAILS ("RTP packet depayloader",
|
|
"Codec/Depayloader/Network",
|
|
"Extracts Speex audio from RTP packets",
|
|
"Edgard Lima <edgard.lima@indt.org.br>");
|
|
|
|
/* RtpSPEEXDepay signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_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\", "
|
|
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
|
|
"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 (GstBaseRTPDepayload * depayload,
|
|
GstBuffer * buf);
|
|
static gboolean gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload,
|
|
GstCaps * caps);
|
|
|
|
GST_BOILERPLATE (GstRtpSPEEXDepay, gst_rtp_speex_depay, GstBaseRTPDepayload,
|
|
GST_TYPE_BASE_RTP_DEPAYLOAD);
|
|
|
|
static void
|
|
gst_rtp_speex_depay_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_rtp_speex_depay_src_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&gst_rtp_speex_depay_sink_template));
|
|
gst_element_class_set_details (element_class, &gst_rtp_speexdepay_details);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
|
|
|
|
gstbasertpdepayload_class->process = gst_rtp_speex_depay_process;
|
|
gstbasertpdepayload_class->set_caps = gst_rtp_speex_depay_setcaps;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay,
|
|
GstRtpSPEEXDepayClass * klass)
|
|
{
|
|
GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay)->clock_rate = 8000;
|
|
}
|
|
|
|
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 (GstBaseRTPDepayload * depayload, GstCaps * caps)
|
|
{
|
|
GstStructure *structure;
|
|
GstRtpSPEEXDepay *rtpspeexdepay;
|
|
gint clock_rate, nb_channels;
|
|
GstBuffer *buf;
|
|
guint8 *data;
|
|
const gchar *params;
|
|
|
|
rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_get_int (structure, "clock-rate", &clock_rate);
|
|
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);
|
|
data = GST_BUFFER_DATA (buf);
|
|
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_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
|
|
|
|
buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment));
|
|
memcpy (GST_BUFFER_DATA (buf), gst_rtp_speex_comment,
|
|
sizeof (gst_rtp_speex_comment));
|
|
|
|
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
|
|
{
|
|
GstBuffer *outbuf = NULL;
|
|
|
|
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
|
|
GST_BUFFER_SIZE (buf),
|
|
gst_rtp_buffer_get_marker (buf),
|
|
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
|
|
|
|
/* nothing special to be done */
|
|
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
|
|
|
|
return outbuf;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_speex_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpspeexdepay",
|
|
GST_RANK_MARGINAL, GST_TYPE_RTP_SPEEX_DEPAY);
|
|
}
|