gstreamer/gst/rtp/gstrtpamrdepay.c
Wim Taymans f48c4cbe42 ext/amrnb/: Update caps with audio/AMR.
Original commit message from CVS:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c: (gst_amrnbenc_setcaps):
* ext/amrnb/amrnbparse.c:
Update caps with audio/AMR.

* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain),
(gst_rtpamrdec_change_state):
* gst/rtp/gstrtpamrdec.h:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
Dont set FT headers twice, it was already in the encoded
bitstream.

* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
* gst/rtsp/rtspconnection.c: (parse_line):
Cleanups

* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property):
* gst/udp/gstudpsrc.h:
Added caps property, we need this soon to type the buffers.
2005-08-19 12:44:35 +00:00

413 lines
11 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 <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include "gstrtpamrdec.h"
/* elementfactory information */
static GstElementDetails gst_rtp_amrdec_details = {
"RTP packet parser",
"Codec/Parser/Network",
"Extracts MPEG audio from RTP packets",
"Wim Taymans <wim@fluendo.com>"
};
/* RtpAMRDec signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_FREQUENCY
};
/* input is an RTP packet
*
* params see RFC 3267, section 8.1
*/
static GstStaticPadTemplate gst_rtpamrdec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"octet-align = (boolean) TRUE, "
"crc = (boolean) FALSE, "
"robust-sorting = (boolean) FALSE, "
"interleaving = (boolean) FALSE, "
"channels = (int) 1, " "rate = (int) 8000"
/* following options are not needed for a decoder
*
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
"maxptime = (int) [ 20, MAX ], "
"ptime = (int) [ 20, MAX ]"
*/
)
);
static GstStaticPadTemplate gst_rtpamrdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000")
);
static void gst_rtpamrdec_class_init (GstRtpAMRDecClass * klass);
static void gst_rtpamrdec_base_init (GstRtpAMRDecClass * klass);
static void gst_rtpamrdec_init (GstRtpAMRDec * rtpamrdec);
static gboolean gst_rtpamrdec_sink_setcaps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_rtpamrdec_chain (GstPad * pad, GstBuffer * buffer);
static void gst_rtpamrdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtpamrdec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstElementStateReturn gst_rtpamrdec_change_state (GstElement * element);
static GstElementClass *parent_class = NULL;
static GType
gst_rtpamrdec_get_type (void)
{
static GType rtpamrdec_type = 0;
if (!rtpamrdec_type) {
static const GTypeInfo rtpamrdec_info = {
sizeof (GstRtpAMRDecClass),
(GBaseInitFunc) gst_rtpamrdec_base_init,
NULL,
(GClassInitFunc) gst_rtpamrdec_class_init,
NULL,
NULL,
sizeof (GstRtpAMRDec),
0,
(GInstanceInitFunc) gst_rtpamrdec_init,
};
rtpamrdec_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstRtpAMRDec",
&rtpamrdec_info, 0);
}
return rtpamrdec_type;
}
static void
gst_rtpamrdec_base_init (GstRtpAMRDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpamrdec_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpamrdec_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_amrdec_details);
}
static void
gst_rtpamrdec_class_init (GstRtpAMRDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gobject_class->set_property = gst_rtpamrdec_set_property;
gobject_class->get_property = gst_rtpamrdec_get_property;
gstelement_class->change_state = gst_rtpamrdec_change_state;
}
static void
gst_rtpamrdec_init (GstRtpAMRDec * rtpamrdec)
{
GstCaps *srccaps;
rtpamrdec->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_rtpamrdec_src_template), "src");
/* FIXME */
srccaps = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
gst_pad_set_caps (rtpamrdec->srcpad, srccaps);
gst_caps_unref (srccaps);
gst_element_add_pad (GST_ELEMENT (rtpamrdec), rtpamrdec->srcpad);
rtpamrdec->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_rtpamrdec_sink_template), "sink");
gst_pad_set_setcaps_function (rtpamrdec->sinkpad, gst_rtpamrdec_sink_setcaps);
gst_pad_set_chain_function (rtpamrdec->sinkpad, gst_rtpamrdec_chain);
gst_element_add_pad (GST_ELEMENT (rtpamrdec), rtpamrdec->sinkpad);
}
static gboolean
gst_rtpamrdec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstStructure *structure;
GstCaps *srccaps;
GstRtpAMRDec *rtpamrdec;
rtpamrdec = GST_RTP_AMR_DEC (GST_OBJECT_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_boolean (structure, "octet-align",
&rtpamrdec->octet_align))
rtpamrdec->octet_align = FALSE;
/* FIXME, force octect align for now until all elements negotiate
* correctly*/
rtpamrdec->octet_align = TRUE;
if (!gst_structure_get_boolean (structure, "crc", &rtpamrdec->crc))
rtpamrdec->crc = FALSE;
if (rtpamrdec->crc) {
/* crc mode implies octet aligned mode */
rtpamrdec->octet_align = TRUE;
}
if (!gst_structure_get_boolean (structure, "robust-sorting",
&rtpamrdec->robust_sorting))
rtpamrdec->robust_sorting = FALSE;
if (rtpamrdec->robust_sorting) {
/* robust_sorting mode implies octet aligned mode */
rtpamrdec->octet_align = TRUE;
}
if (!gst_structure_get_boolean (structure, "interleaving",
&rtpamrdec->interleaving))
rtpamrdec->interleaving = FALSE;
if (rtpamrdec->interleaving) {
/* interleaving mode implies octet aligned mode */
rtpamrdec->octet_align = TRUE;
}
if (!gst_structure_get_int (structure, "channels", &rtpamrdec->channels))
rtpamrdec->channels = 1;
if (!gst_structure_get_int (structure, "rate", &rtpamrdec->rate))
rtpamrdec->rate = 8000;
/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving for now */
if (rtpamrdec->channels != 1)
return FALSE;
if (rtpamrdec->rate != 8000)
return FALSE;
if (rtpamrdec->octet_align != TRUE)
return FALSE;
if (rtpamrdec->crc != FALSE)
return FALSE;
if (rtpamrdec->robust_sorting != FALSE)
return FALSE;
if (rtpamrdec->interleaving != FALSE)
return FALSE;
srccaps = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, rtpamrdec->channels,
"rate", G_TYPE_INT, rtpamrdec->rate, NULL);
gst_pad_set_caps (rtpamrdec->srcpad, srccaps);
gst_caps_unref (srccaps);
rtpamrdec->negotiated = TRUE;
return TRUE;
}
static GstFlowReturn
gst_rtpamrdec_chain (GstPad * pad, GstBuffer * buf)
{
GstRtpAMRDec *rtpamrdec;
GstBuffer *outbuf;
GstFlowReturn ret;
rtpamrdec = GST_RTP_AMR_DEC (GST_OBJECT_PARENT (pad));
if (!rtpamrdec->negotiated)
goto not_negotiated;
if (!gst_rtpbuffer_validate (buf))
goto bad_packet;
/* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving data is to be parsed */
{
gint payload_len;
guint8 *payload;
guint32 timestamp;
guint8 CMR, F, FT, Q;
payload_len = gst_rtpbuffer_get_payload_len (buf);
/* need at least 2 bytes for the header */
if (payload_len < 2)
goto bad_packet;
payload = gst_rtpbuffer_get_payload (buf);
/* parse header
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+..
* | CMR=6 |R|R|R|R|0|FT#1=5 |Q|P|P|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+..
*/
CMR = (payload[0] & 0xf0) >> 4;
F = (payload[1] & 0x80) >> 7;
/* we only support 1 packet per RTP packet for now */
if (F != 0)
goto one_packet_only;
FT = (payload[1] & 0x78) >> 3;
Q = (payload[1] & 0x04) >> 2;
/* skip packet */
if (FT > 9 && FT < 15) {
ret = GST_FLOW_OK;
goto skip;
}
/* strip header now, leave FT in the data for the decoder */
payload_len -= 1;
payload += 1;
timestamp = gst_rtpbuffer_get_timestamp (buf);
outbuf = gst_buffer_new_and_alloc (payload_len);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp * GST_SECOND / 8000;
memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (rtpamrdec->srcpad));
GST_DEBUG ("gst_rtpamrdec_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_pad_push (rtpamrdec->srcpad, outbuf);
skip:
gst_buffer_unref (buf);
}
return ret;
not_negotiated:
{
GST_DEBUG ("not_negotiated");
gst_buffer_unref (buf);
return GST_FLOW_NOT_NEGOTIATED;
}
bad_packet:
{
GST_DEBUG ("Packet did not validate");
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
one_packet_only:
{
GST_DEBUG ("One packet per RTP packet only");
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static void
gst_rtpamrdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpAMRDec *rtpamrdec;
rtpamrdec = GST_RTP_AMR_DEC (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtpamrdec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRtpAMRDec *rtpamrdec;
rtpamrdec = GST_RTP_AMR_DEC (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstElementStateReturn
gst_rtpamrdec_change_state (GstElement * element)
{
GstRtpAMRDec *rtpamrdec;
gint transition;
GstElementStateReturn ret;
rtpamrdec = GST_RTP_AMR_DEC (element);
transition = GST_STATE_TRANSITION (element);
switch (transition) {
case GST_STATE_NULL_TO_READY:
break;
case GST_STATE_READY_TO_PAUSED:
/* FIXME, don't require negotiation until all elements
* do */
rtpamrdec->negotiated = TRUE;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
switch (transition) {
case GST_STATE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_rtpamrdec_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpamrdec",
GST_RANK_NONE, GST_TYPE_RTP_AMR_DEC);
}