gstreamer/gst/rtp/gstrtpamrpay.c
Philippe Kalaf 9a6ca70be2 gst-plugins-good/gst/rtp/: Fixed payload range in payloder caps. Removed payload range completly from depayloaders as...
Original commit message from CVS:
2005-12-14  Philippe Khalaf  <burger@speedy.org>

* gst-plugins-good/gst/rtp/gstasteriskh263.c:
* gst-plugins-good/gst/rtp/gstrtpamrdepay.c:
* gst-plugins-good/gst/rtp/gstrtpamrpay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
* gst-plugins-good/gst/rtp/gstrtpgsmdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263pay.c:
* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
* gst-plugins-good/gst/rtp/gstrtph263ppay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
* gst-plugins-good/gst/rtp/gstrtpmp4vpay.c:
* gst-plugins-good/gst/rtp/gstrtpmpadepay.c:
* gst-plugins-good/gst/rtp/gstrtpmpapay.c:
* gst-plugins-good/gst/rtp/README:
Fixed payload range in payloder caps. Removed payload range completly from
depayloaders as they don't require payload type in their caps. In effect,
there isn't any specific payload type for any given codec, only suggestions.
Fixes bug #324011.
2005-12-14 18:07:16 +00:00

225 lines
6.3 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpamrpay.h"
/* references:
*
* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive
* Multi-Rate Wideband (AMR-WB) Audio Codecs.
*/
/* elementfactory information */
static GstElementDetails gst_rtp_amrpay_details = {
"RTP packet parser",
"Codec/Payloader/Network",
"Payode AMR audio into RTP packets (RFC 3267)",
"Wim Taymans <wim@fluendo.com>"
};
static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000")
);
static GstStaticPadTemplate gst_rtp_amr_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) [ 96, 127 ], "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"AMR\", "
"encoding-params = (string) \"1\", "
"octet-align = (string) \"1\", "
"crc = (string) \"0\", "
"robust-sorting = (string) \"0\", "
"interleaving = (string) \"0\", "
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
"mode-change-neighbor = (string) { \"0\", \"1\" }, "
"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
);
static void gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass);
static void gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass);
static void gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay);
static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_amr_pay_get_type (void)
{
static GType rtpamrpay_type = 0;
if (!rtpamrpay_type) {
static const GTypeInfo rtpamrpay_info = {
sizeof (GstRtpAMRPayClass),
(GBaseInitFunc) gst_rtp_amr_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_amr_pay_class_init,
NULL,
NULL,
sizeof (GstRtpAMRPay),
0,
(GInstanceInitFunc) gst_rtp_amr_pay_init,
};
rtpamrpay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMRPay",
&rtpamrpay_info, 0);
}
return rtpamrpay_type;
}
static void
gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_amr_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_amr_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_amrpay_details);
}
static void
gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * 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_rtp_amr_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
}
static void
gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay)
{
}
static gboolean
gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpAMRPay *rtpamrpay;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
/* don't set the defaults
*
* "crc", G_TYPE_STRING, "0",
* "robust-sorting", G_TYPE_STRING, "0",
* "interleaving", G_TYPE_STRING, "0",
*/
NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpAMRPay *rtpamrpay;
GstFlowReturn ret;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one AMR frame per RTP packet for now,
* octet aligned, no interleaving, single channel, no CRC,
* no robust-sorting. */
/* we need one extra byte for the CMR, the ToC is in the input
* data */
payload_len = size + 1;
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (GST_BUFFER_SIZE (outbuf) < GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay));
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* | CMR |R|R|R|R|
* +-+-+-+-+-+-+-+-+
*/
payload[0] = 0xF0; /* CMR, no specific mode requested */
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[1], data, size);
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |F| FT |Q|P|P|
* +-+-+-+-+-+-+-+-+
*/
/* clear F flag */
payload[1] = payload[1] & 0x7f;
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
gboolean
gst_rtp_amr_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpamrpay",
GST_RANK_NONE, GST_TYPE_RTP_AMR_PAY);
}