gstreamer/gst/rtp/gstrtpgsmenc.c
Stéphane Loeuillet 61a021fbba ext/mad/gstid3tag.c: move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
Original commit message from CVS:
* ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
* gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"

* gst/auparse/gstauparse.c :
- add code (commented for now) to support audio/x-adpcm on src pad
(we have no decoder for those layout yet)

* gst/cdxaparse/gstcdxaparse.c :
* gst/cdxaparse/gstcdxaparse.h :
- partial rewrite using RiffRead (ripped iain's wavparse code)

* gst/rtp/gstrtpL16enc.c : typo
* gst/rtp/gstrtpgsmenc.c : typo
2004-05-13 21:27:14 +00:00

332 lines
9.1 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 <stdlib.h>
#include <string.h>
#include "gstrtpgsmenc.h"
/* elementfactory information */
static GstElementDetails gst_rtpgsmenc_details = {
"RTP GSM Audio Encoder",
"Codec/Encoder/Network",
"Encodes GSM audio into a RTP packet",
"Zeeshan Ali <zak147@yahoo.com>"
};
/* RtpGSMEnc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
/* FILL ME */
ARG_0,
};
static GstStaticPadTemplate gst_rtpgsmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) [ 1000, 48000 ]")
);
static GstStaticPadTemplate gst_rtpgsmenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static void gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass);
static void gst_rtpgsmenc_base_init (GstRtpGSMEncClass * klass);
static void gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc);
static void gst_rtpgsmenc_chain (GstPad * pad, GstData * _data);
static void gst_rtpgsmenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtpgsmenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstPadLinkReturn gst_rtpgsmenc_sinkconnect (GstPad * pad,
const GstCaps * caps);
static GstElementStateReturn gst_rtpgsmenc_change_state (GstElement * element);
static GstElementClass *parent_class = NULL;
static GType
gst_rtpgsmenc_get_type (void)
{
static GType rtpgsmenc_type = 0;
if (!rtpgsmenc_type) {
static const GTypeInfo rtpgsmenc_info = {
sizeof (GstRtpGSMEncClass),
(GBaseInitFunc) gst_rtpgsmenc_base_init,
NULL,
(GClassInitFunc) gst_rtpgsmenc_class_init,
NULL,
NULL,
sizeof (GstRtpGSMEnc),
0,
(GInstanceInitFunc) gst_rtpgsmenc_init,
};
rtpgsmenc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstRtpGSMEnc",
&rtpgsmenc_info, 0);
}
return rtpgsmenc_type;
}
static void
gst_rtpgsmenc_base_init (GstRtpGSMEncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpgsmenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpgsmenc_src_template));
gst_element_class_set_details (element_class, &gst_rtpgsmenc_details);
}
static void
gst_rtpgsmenc_class_init (GstRtpGSMEncClass * 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_rtpgsmenc_set_property;
gobject_class->get_property = gst_rtpgsmenc_get_property;
gstelement_class->change_state = gst_rtpgsmenc_change_state;
}
static void
gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc)
{
rtpgsmenc->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_rtpgsmenc_sink_template), "sink");
rtpgsmenc->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_rtpgsmenc_sink_template), "src");
gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->sinkpad);
gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->srcpad);
gst_pad_set_chain_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_chain);
gst_pad_set_link_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_sinkconnect);
rtpgsmenc->frequency = 8000;
rtpgsmenc->next_time = 0;
rtpgsmenc->time_interval = 0;
rtpgsmenc->seq = 0;
rtpgsmenc->ssrc = random ();
}
static GstPadLinkReturn
gst_rtpgsmenc_sinkconnect (GstPad * pad, const GstCaps * caps)
{
GstRtpGSMEnc *rtpgsmenc;
GstStructure *structure;
gboolean ret;
rtpgsmenc = GST_RTP_GSM_ENC (gst_pad_get_parent (pad));
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &rtpgsmenc->frequency);
if (!ret)
return GST_PAD_LINK_REFUSED;
/* Pre-calculate what we can */
rtpgsmenc->time_interval = GST_SECOND / (2 * rtpgsmenc->frequency);
return GST_PAD_LINK_OK;
}
void
gst_rtpgsmenc_htons (GstBuffer * buf)
{
gint16 *i, *len;
/* FIXME: is this code correct or even sane at all? */
i = (gint16 *) GST_BUFFER_DATA (buf);
len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *);
for (; i < len; i++) {
*i = g_htons (*i);
}
}
static void
gst_rtpgsmenc_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
GstRtpGSMEnc *rtpgsmenc;
GstBuffer *outbuf;
Rtp_Packet packet;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
rtpgsmenc = GST_RTP_GSM_ENC (GST_OBJECT_PARENT (pad));
g_return_if_fail (rtpgsmenc != NULL);
g_return_if_fail (GST_IS_RTP_GSM_ENC (rtpgsmenc));
if (GST_IS_EVENT (buf)) {
GstEvent *event = GST_EVENT (buf);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_DISCONTINUOUS:
GST_DEBUG ("discont");
rtpgsmenc->next_time = 0;
gst_pad_event_default (pad, event);
return;
default:
gst_pad_event_default (pad, event);
return;
}
}
/* We only need the header */
packet = rtp_packet_new_allocate (0, 0, 0);
rtp_packet_set_csrc_count (packet, 0);
rtp_packet_set_extension (packet, 0);
rtp_packet_set_padding (packet, 0);
rtp_packet_set_version (packet, RTP_VERSION);
rtp_packet_set_marker (packet, 0);
rtp_packet_set_ssrc (packet, g_htonl (rtpgsmenc->ssrc));
rtp_packet_set_seq (packet, g_htons (rtpgsmenc->seq));
rtp_packet_set_timestamp (packet,
g_htonl ((guint32) rtpgsmenc->next_time / GST_SECOND));
rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_GSM);
/* FIXME: According to RFC 1890, this is required, right? */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
gst_rtpgsmenc_htons (buf);
#endif
outbuf = gst_buffer_new ();
GST_BUFFER_SIZE (outbuf) =
rtp_packet_get_packet_len (packet) + GST_BUFFER_SIZE (buf);
GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf));
GST_BUFFER_TIMESTAMP (outbuf) = rtpgsmenc->next_time;
memcpy (GST_BUFFER_DATA (outbuf), packet->data,
rtp_packet_get_packet_len (packet));
memcpy (GST_BUFFER_DATA (outbuf) + rtp_packet_get_packet_len (packet),
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
GST_DEBUG ("gst_rtpgsmenc_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
gst_pad_push (rtpgsmenc->srcpad, GST_DATA (outbuf));
++rtpgsmenc->seq;
rtpgsmenc->next_time += rtpgsmenc->time_interval * GST_BUFFER_SIZE (buf);
rtp_packet_free (packet);
gst_buffer_unref (buf);
}
static void
gst_rtpgsmenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpGSMEnc *rtpgsmenc;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_RTP_GSM_ENC (object));
rtpgsmenc = GST_RTP_GSM_ENC (object);
switch (prop_id) {
default:
break;
}
}
static void
gst_rtpgsmenc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRtpGSMEnc *rtpgsmenc;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_RTP_GSM_ENC (object));
rtpgsmenc = GST_RTP_GSM_ENC (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstElementStateReturn
gst_rtpgsmenc_change_state (GstElement * element)
{
GstRtpGSMEnc *rtpgsmenc;
g_return_val_if_fail (GST_IS_RTP_GSM_ENC (element), GST_STATE_FAILURE);
rtpgsmenc = GST_RTP_GSM_ENC (element);
GST_DEBUG ("state pending %d\n", GST_STATE_PENDING (element));
/* if going down into NULL state, close the file if it's open */
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
break;
case GST_STATE_READY_TO_NULL:
break;
default:
break;
}
/* if we haven't failed already, give the parent class a chance to ;-) */
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
gboolean
gst_rtpgsmenc_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpgsmenc",
GST_RANK_NONE, GST_TYPE_RTP_GSM_ENC);
}