gstreamer/ext/gsm/gstgsmenc.c

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/* 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 <string.h>
#include "gstgsmenc.h"
/* elementfactory information */
GstElementDetails gst_gsmenc_details = {
"GSM audio encoder",
"Codec/Encoder/Audio",
"Encodes audio using GSM",
"Wim Taymans <wim.taymans@chello.be>",
};
/* GSMEnc signals and args */
enum
{
FRAME_ENCODED,
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
/* FILL ME */
};
static void gst_gsmenc_base_init (gpointer g_class);
static void gst_gsmenc_class_init (GstGSMEnc * klass);
static void gst_gsmenc_init (GstGSMEnc * gsmenc);
static GstFlowReturn gst_gsmenc_chain (GstPad * pad, GstBuffer * buf);
static GstElementClass *parent_class = NULL;
static guint gst_gsmenc_signals[LAST_SIGNAL] = { 0 };
GType
gst_gsmenc_get_type (void)
{
static GType gsmenc_type = 0;
if (!gsmenc_type) {
static const GTypeInfo gsmenc_info = {
sizeof (GstGSMEncClass),
gst_gsmenc_base_init,
NULL,
(GClassInitFunc) gst_gsmenc_class_init,
NULL,
NULL,
sizeof (GstGSMEnc),
0,
(GInstanceInitFunc) gst_gsmenc_init,
};
gsmenc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstGSMEnc", &gsmenc_info, 0);
}
return gsmenc_type;
}
static GstStaticPadTemplate gsmenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
);
static GstStaticPadTemplate gsmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
);
static void
gst_gsmenc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmenc_src_template));
gst_element_class_set_details (element_class, &gst_gsmenc_details);
}
static void
gst_gsmenc_class_init (GstGSMEnc * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gst_gsmenc_signals[FRAME_ENCODED] =
g_signal_new ("frame-encoded", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstGSMEncClass, frame_encoded), NULL,
NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
}
static void
gst_gsmenc_init (GstGSMEnc * gsmenc)
{
/* create the sink and src pads */
gsmenc->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gsmenc_sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->sinkpad);
gst_pad_set_chain_function (gsmenc->sinkpad, gst_gsmenc_chain);
gsmenc->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gsmenc_src_template), "src");
gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->srcpad);
gsmenc->state = gsm_create ();
gsmenc->bufsize = 0;
gsmenc->next_ts = 0;
}
static GstFlowReturn
gst_gsmenc_chain (GstPad * pad, GstBuffer * buf)
{
GstGSMEnc *gsmenc;
gsmenc = GST_GSMENC (GST_OBJECT_PARENT (pad));
if (GST_IS_EVENT (buf)) {
GstEvent *event = GST_EVENT (buf);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:{
gst_pad_push_event (gsmenc->srcpad, gst_event_new_eos ());
gst_pad_push (gsmenc->srcpad, buf);
break;
}
case GST_EVENT_NEWSEGMENT:{
/* drop the discontinuity */
break;
}
default:{
gst_pad_event_default (pad, event);
break;
}
}
return GST_FLOW_OK;
} else if (GST_IS_BUFFER (buf)) {
gsm_signal *data;
guint size;
GstCaps *tempcaps = NULL;
data = (gsm_signal *) GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf) / sizeof (gsm_signal);
if (gsmenc->bufsize && (gsmenc->bufsize + size >= 160)) {
GstBuffer *outbuf;
memcpy (gsmenc->buffer + gsmenc->bufsize, data,
(160 - gsmenc->bufsize) * sizeof (gsm_signal));
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
gsmenc->next_ts += 20 * GST_MSECOND;
gsm_encode (gsmenc->state, gsmenc->buffer,
(gsm_byte *) GST_BUFFER_DATA (outbuf));
tempcaps = gst_caps_new_simple ("audio/x-gsm",
"rate", G_TYPE_INT, 8000, "channels", G_TYPE_INT, 1, NULL);
gst_buffer_set_caps (outbuf, tempcaps);
gst_pad_push (gsmenc->srcpad, outbuf);
size -= (160 - gsmenc->bufsize);
data += (160 - gsmenc->bufsize);
gsmenc->bufsize = 0;
}
while (size >= 160) {
GstBuffer *outbuf;
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
gsmenc->next_ts += 20 * GST_MSECOND;
gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
/* I was wondering that gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmenc->srcpad));
* could work, but it doens't work */
tempcaps = gst_caps_new_simple ("audio/x-gsm",
"rate", G_TYPE_INT, 8000, "channels", G_TYPE_INT, 1, NULL);
gst_buffer_set_caps (outbuf, tempcaps);
gst_pad_push (gsmenc->srcpad, outbuf);
size -= 160;
data += 160;
}
if (size) {
memcpy (gsmenc->buffer + gsmenc->bufsize, data,
size * sizeof (gsm_signal));
gsmenc->bufsize += size;
}
/*gst_buffer_unref (buf); */
return GST_FLOW_OK;
}
return GST_FLOW_OK;
}