gstreamer/gst/rtp/gstrtpilbcpay.c
Philippe Kalaf 07f9b4f658 gst/rtp/: Ported mulaw and alaw payloaders to use new base class
Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpilbcpay.h:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcdepay.h:
Added new iLBC payloader/depayloader. Payloader uses new audio payload base
class.
2006-04-13 03:42:51 +00:00

161 lines
5 KiB
C

/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* 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 "gstrtpilbcpay.h"
#include <gst/rtp/gstrtpbuffer.h>
/* elementfactory information */
static GstElementDetails gst_rtpilbcpay_details = {
"RTP Payloader for iLBC Audio",
"Codec/Payloader/Network",
"Packetize iLBC audio streams into RTP packets",
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>"
};
GST_DEBUG_CATEGORY (rtpilbcpay_debug);
#define GST_CAT_DEFAULT (rtpilbcpay_debug)
static GstStaticPadTemplate gst_rtpilbcpay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-iLBC, " "mode = (int) {20, 30}")
);
static GstStaticPadTemplate gst_rtpilbcpay_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) \"iLBC\", " "mode = (int) {20, 30}")
);
static gboolean gst_rtpilbcpay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRTPILBCPay, gst_rtpilbcpay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtpilbcpay_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_rtpilbcpay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpilbcpay_src_template));
gst_element_class_set_details (element_class, &gst_rtpilbcpay_details);
}
static void
gst_rtpilbcpay_class_init (GstRTPILBCPayClass * 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_rtpilbcpay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpilbcpay_debug, "rtpilbcpay", 0,
"iLBC audio RTP payloader");
}
static void
gst_rtpilbcpay_init (GstRTPILBCPay * rtpilbcpay, GstRTPILBCPayClass * klass)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertppayload = GST_BASE_RTP_PAYLOAD (rtpilbcpay);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpilbcpay);
/* we don't set the payload type, it should be set by the application using
* the pt property or the default 96 will be used */
basertppayload->clock_rate = 8000;
rtpilbcpay->mode = -1;
/* tell basertpaudiopayload that this is a frame based codec */
gst_basertpaudiopayload_set_frame_based (basertpaudiopayload);
}
static gboolean
gst_rtpilbcpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
{
GstRTPILBCPay *rtpilbcpay;
GstBaseRTPAudioPayload *basertpaudiopayload;
gboolean ret;
gint mode;
GstStructure *structure;
const char *payload_name;
rtpilbcpay = GST_RTP_ILBC_PAY (basertppayload);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "mode", &mode);
if (mode != 20 && mode != 30) {
return FALSE;
}
payload_name = gst_structure_get_name (structure);
if (g_strcasecmp ("audio/x-iLBC", payload_name) == 0) {
gst_basertppayload_set_options (basertppayload, "audio", TRUE, "iLBC",
8000);
/* set options for this frame based audio codec */
gst_basertpaudiopayload_set_frame_options (basertpaudiopayload,
mode, mode == 30 ? 50 : 38);
} else {
return FALSE;
}
ret =
gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_INT, mode,
NULL);
if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1) {
GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! \
Mode cannot change while streaming", rtpilbcpay->mode, mode);
return FALSE;
}
rtpilbcpay->mode = mode;
return ret;
}
gboolean
gst_rtp_ilbc_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpilbcpay",
GST_RANK_NONE, GST_TYPE_RTP_ILBC_PAY);
}