gstreamer/gst/rtp/gstrtpg726pay.c
Wim Taymans 5e27695ca2 gst/rtp/: Fix the descriptions and fix some email addresses.
Original commit message from CVS:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstasteriskh263.h:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpL16pay.h:
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps):
* gst/rtp/gstrtpac3depay.h:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrdepay.h:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpamrpay.h:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpdepay.h:
* gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps):
* gst/rtp/gstrtpg726depay.c:
* gst/rtp/gstrtpg726pay.c:
* gst/rtp/gstrtpg729depay.c:
* gst/rtp/gstrtpg729pay.c:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps):
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps):
* gst/rtp/gstrtph263depay.h:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pay.h:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
* gst/rtp/gstrtph263pdepay.h:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtph263ppay.h:
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtph264depay.h:
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtph264pay.h:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpjpegdepay.h:
* gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps):
* gst/rtp/gstrtpmp1sdepay.h:
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
* gst/rtp/gstrtpmp2tdepay.h:
* gst/rtp/gstrtpmp2tpay.c:
* gst/rtp/gstrtpmp2tpay.h:
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps):
* gst/rtp/gstrtpmp4apay.c:
* gst/rtp/gstrtpmp4apay.h:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps):
* gst/rtp/gstrtpmp4gdepay.h:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps):
* gst/rtp/gstrtpmp4vdepay.h:
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event):
* gst/rtp/gstrtpmp4vpay.h:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpadepay.h:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtpmpapay.h:
* gst/rtp/gstrtpmpvdepay.c:
* gst/rtp/gstrtpmpvdepay.h:
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtpsv3vdepay.h:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheoradepay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtptheorapay.h:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
* gst/rtp/gstrtpvorbispay.h:
* gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps):
* gst/rtp/gstrtpvrawpay.c:
Fix the descriptions and fix some email addresses.
2008-11-25 18:03:02 +00:00

166 lines
5.1 KiB
C

/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
* Copyright (C) 2005 Nokia Corporation <kai.vehmanen@nokia.com>
* Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.com>
*
* 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 <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpg726pay.h"
static const GstElementDetails gst_rtp_g726_pay_details =
GST_ELEMENT_DETAILS ("RTP G.726 payloader",
"Codec/Payloader/Network",
"Payload-encodes G.726 audio into a RTP packet",
"Axis Communications <dev-gstreamer@axis.com>");
static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-adpcm, "
"channels = (int) 1, "
"rate = (int) 8000, "
"bitrate = (int) { 16000, 24000, 32000, 40000 }, "
"layout = (string) \"g726\"")
);
static GstStaticPadTemplate gst_rtp_g726_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\" } ")
);
static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_g726_pay_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_g726_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_g726_pay_details);
}
static void
gst_rtp_g726_pay_class_init (GstRtpG726PayClass * 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_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
}
static void
gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg726pay);
GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000;
/* sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
}
static gboolean
gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gchar *encoding_name;
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstBaseRTPAudioPayload *basertpaudiopayload;
gint bitrate;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
if (!gst_structure_get_int (structure, "bitrate", &bitrate))
bitrate = 32000;
switch (bitrate) {
case 16000:
encoding_name = g_strdup ("G726-16");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
2);
break;
case 24000:
encoding_name = g_strdup ("G726-24");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
3);
break;
case 32000:
encoding_name = g_strdup ("G726-32");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
4);
break;
case 40000:
encoding_name = g_strdup ("G726-40");
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
5);
break;
default:
goto invalid_bitrate;
}
gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
gst_basertppayload_set_outcaps (payload, NULL);
g_free (encoding_name);
return TRUE;
/* ERRORS */
invalid_bitrate:
{
GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", bitrate);
return FALSE;
}
}
gboolean
gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg726pay",
GST_RANK_NONE, GST_TYPE_RTP_G726_PAY);
}