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/* 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 "gstrtpamrenc.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_amrenc_details = {
"RTP packet parser",
"Codec/Encoder/Network",
"Encode AMR audio into RTP packets (RFC 3267)",
"Wim Taymans <wim@fluendo.com>"
};
static GstStaticPadTemplate gst_rtpamrenc_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_rtpamrenc_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, 255 ], "
"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_rtpamrenc_class_init (GstRtpAMREncClass * klass);
static void gst_rtpamrenc_base_init (GstRtpAMREncClass * klass);
static void gst_rtpamrenc_init (GstRtpAMREnc * rtpamrenc);
static gboolean gst_rtpamrenc_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtpamrenc_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtpamrenc_get_type (void)
{
static GType rtpamrenc_type = 0;
if (!rtpamrenc_type) {
static const GTypeInfo rtpamrenc_info = {
sizeof (GstRtpAMREncClass),
(GBaseInitFunc) gst_rtpamrenc_base_init,
NULL,
(GClassInitFunc) gst_rtpamrenc_class_init,
NULL,
NULL,
sizeof (GstRtpAMREnc),
0,
(GInstanceInitFunc) gst_rtpamrenc_init,
};
rtpamrenc_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMREnc",
&rtpamrenc_info, 0);
}
return rtpamrenc_type;
}
static void
gst_rtpamrenc_base_init (GstRtpAMREncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpamrenc_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpamrenc_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_amrenc_details);
}
static void
gst_rtpamrenc_class_init (GstRtpAMREncClass * 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_rtpamrenc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpamrenc_handle_buffer;
}
static void
gst_rtpamrenc_init (GstRtpAMREnc * rtpamrenc)
{
}
static gboolean
gst_rtpamrenc_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpAMREnc *rtpamrenc;
rtpamrenc = GST_RTP_AMR_ENC (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_rtpamrenc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpAMREnc *rtpamrenc;
GstFlowReturn ret;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
rtpamrenc = GST_RTP_AMR_ENC (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_rtpbuffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (GST_BUFFER_SIZE (outbuf) < GST_BASE_RTP_PAYLOAD_MTU (rtpamrenc));
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtpbuffer_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_rtpamrenc_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpamrenc",
GST_RANK_NONE, GST_TYPE_RTP_AMR_ENC);
}
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