gstreamer/gst/rtp/gstrtpamrpay.c
Tim-Philipp Müller 05eaedc496 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) plus two minor macro fixes.
Original commit message from CVS:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/annodex/gstcmmlparser.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/gdk_pixbuf/pixbufscale.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalphacolor.c:
* gst/cutter/gstcutter.c:
* gst/debug/gstnavigationtest.c:
* gst/icydemux/gsticydemux.c:
* gst/level/gstlevel.c:
* gst/multipart/multipart.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstvideoflip.c:
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
plus two minor macro fixes.
2006-06-22 19:31:04 +00:00

329 lines
9.1 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 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 <gst/rtp/gstrtpbuffer.h>
#include "gstrtpamrpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
#define GST_CAT_DEFAULT (rtpamrpay_debug)
/* 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 const GstElementDetails gst_rtp_amrpay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Payloader/Network",
"Payload-encode 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_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_amr_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
"AMR RTP Payloader");
}
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;
}
/* -1 is invalid */
static gint frame_size[16] = {
12, 13, 15, 17, 19, 20, 26, 31,
5, -1, -1, -1, -1, -1, -1, 0
};
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, *payload_amr;
GstClockTime timestamp;
guint packet_len, mtu;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only
* octet aligned, no interleaving, single channel, no CRC,
* no robust-sorting. */
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
/* first count number of packets and total amr frame size */
amr_len = num_packets = num_nonempty_packets = 0;
for (i = 0; i < size; i++) {
guint8 FT;
gint fr_size;
FT = (data[i] & 0x78) >> 3;
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (basepayload, "frame size %d", fr_size);
/* FIXME, we don't handle this yet.. */
if (fr_size <= 0)
goto wrong_size;
amr_len += fr_size;
num_nonempty_packets++;
num_packets++;
i += fr_size;
}
if (amr_len > size)
goto incomplete_frame;
/* we need one extra byte for the CMR, the ToC is in the input
* data */
payload_len = size + 1;
/* get packet len to check against MTU */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
if (packet_len > mtu)
goto too_big;
/* now alloc output buffer */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy timestamp, or fabricate one */
if (timestamp != GST_CLOCK_TIME_NONE)
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
else {
/* AMR (nb) and AMR-WB both have 20 ms per frame */
/* FIXME: when we do more than one AMR frame per packet, fix this */
gint count = basepayload->seqnum - basepayload->seqnum_base;
GST_BUFFER_TIMESTAMP (outbuf) = count * 20 * GST_MSECOND;
}
/* get payload, this is now writable */
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 */
/* this is where we copy the AMR data, after num_packets FTs and the
* CMR. */
payload_amr = payload + num_packets + 1;
/* copy data in payload, first we copy all the FTs then all
* the AMR data. The last FT has to have the F flag cleared. */
for (i = 1; i <= num_packets; i++) {
guint8 FT;
gint fr_size;
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |F| FT |Q|P|P| more FT...
* +-+-+-+-+-+-+-+-+
*/
FT = (*data & 0x78) >> 3;
fr_size = frame_size[FT];
if (i == num_packets)
/* last packet, clear F flag */
payload[i] = *data & 0x7f;
else
/* set F flag */
payload[i] = *data | 0x80;
memcpy (payload_amr, &data[1], fr_size);
/* all sizes are > 0 since we checked for that above */
data += fr_size + 1;
payload_amr += fr_size;
}
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received AMR frame with size <= 0"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
incomplete_frame:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received incomplete AMR frames"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
too_big:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received too many AMR frames for MTU"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
gboolean
gst_rtp_amr_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpamrpay",
GST_RANK_NONE, GST_TYPE_RTP_AMR_PAY);
}