gstreamer/ext/amrnb/amrnbdec.c
Ronald S. Bultje 56675a8221 Add support for AMR-NB (mobile phone audio format; #155163, #163286).
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/amrnb/Makefile.am:
* ext/amrnb/amrnb.c: (plugin_init):
* ext/amrnb/amrnbdec.c: (gst_amrnbdec_get_type),
(gst_amrnbdec_base_init), (gst_amrnbdec_class_init),
(gst_amrnbdec_init), (gst_amrnbdec_link), (gst_amrnbdec_chain),
(gst_amrnbdec_state_change):
* ext/amrnb/amrnbdec.h:
* ext/amrnb/amrnbparse.c: (gst_amrnbparse_get_type),
(gst_amrnbparse_base_init), (gst_amrnbparse_class_init),
(gst_amrnbparse_init), (gst_amrnbparse_formats),
(gst_amrnbparse_querytypes), (gst_amrnbparse_query),
(gst_amrnbparse_handle_event), (gst_amrnbparse_reserve),
(gst_amrnbparse_loop), (gst_amrnbparse_state_change):
* ext/amrnb/amrnbparse.h:
Add support for AMR-NB (mobile phone audio format; #155163, #163286).
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add AMR-NB/-WB raw formats.
* ext/alsa/gstalsa.c: (gst_alsa_link):
Keep valid time when changing format.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
Add some more format-specific options (#140141, #143555, #155163).
2005-01-28 10:36:12 +00:00

222 lines
6.6 KiB
C

/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* 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 "amrnbdec.h"
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-amr-nb, "
"rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 2 ]")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 16, "
"depth = (int) 16, "
"signed = (boolean) TRUE, "
"endianness = (int) BYTE_ORDER, "
"rate = (int) [ 1000, 96000 ]," "channels = (int) [ 1, 2 ]")
);
static void gst_amrnbdec_base_init (GstAmrnbDecClass * klass);
static void gst_amrnbdec_class_init (GstAmrnbDecClass * klass);
static void gst_amrnbdec_init (GstAmrnbDec * amrnbdec);
static void gst_amrnbdec_chain (GstPad * pad, GstData * data);
static GstPadLinkReturn gst_amrnbdec_link (GstPad * pad, const GstCaps * caps);
static GstElementStateReturn gst_amrnbdec_state_change (GstElement * element);
static GstElementClass *parent_class = NULL;
GType
gst_amrnbdec_get_type (void)
{
static GType amrnbdec_type = 0;
if (!amrnbdec_type) {
static const GTypeInfo amrnbdec_info = {
sizeof (GstAmrnbDecClass),
(GBaseInitFunc) gst_amrnbdec_base_init,
NULL,
(GClassInitFunc) gst_amrnbdec_class_init,
NULL,
NULL,
sizeof (GstAmrnbDec),
0,
(GInstanceInitFunc) gst_amrnbdec_init,
};
amrnbdec_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstAmrnbDec", &amrnbdec_info, 0);
}
return amrnbdec_type;
}
static void
gst_amrnbdec_base_init (GstAmrnbDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstElementDetails gst_amrnbdec_details = {
"AMR-NB decoder",
"Codec/Decoder/Audio",
"Adaptive Multi-Rate Narrow-Band audio decoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>"
};
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details (element_class, &gst_amrnbdec_details);
}
static void
gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
element_class->change_state = gst_amrnbdec_state_change;
}
static void
gst_amrnbdec_init (GstAmrnbDec * amrnbdec)
{
/* create the sink pad */
amrnbdec->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get (&sink_template),
"sink");
gst_pad_set_link_function (amrnbdec->sinkpad, gst_amrnbdec_link);
gst_pad_set_chain_function (amrnbdec->sinkpad, gst_amrnbdec_chain);
gst_element_add_pad (GST_ELEMENT (amrnbdec), amrnbdec->sinkpad);
/* create the src pad */
amrnbdec->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get (&src_template),
"src");
gst_pad_use_explicit_caps (amrnbdec->srcpad);
gst_element_add_pad (GST_ELEMENT (amrnbdec), amrnbdec->srcpad);
/* init rest */
amrnbdec->handle = NULL;
amrnbdec->channels = 0;
amrnbdec->rate = 0;
amrnbdec->ts = 0;
}
static GstPadLinkReturn
gst_amrnbdec_link (GstPad * pad, const GstCaps * caps)
{
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstAmrnbDec *amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
GstCaps *copy;
/* get channel count */
gst_structure_get_int (structure, "channels", &amrnbdec->channels);
gst_structure_get_int (structure, "rate", &amrnbdec->rate);
/* create reverse caps */
copy = gst_caps_new_simple ("audio/x-raw-int",
"channels", G_TYPE_INT, amrnbdec->channels,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", G_TYPE_INT, amrnbdec->rate, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!gst_pad_set_explicit_caps (amrnbdec->srcpad, copy))
return GST_PAD_LINK_REFUSED;
return GST_PAD_LINK_OK;
}
static void
gst_amrnbdec_chain (GstPad * pad, GstData * in_data)
{
const gint block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5,
0, 0, 0, 0, 0, 0, 0
};
GstAmrnbDec *amrnbdec = GST_AMRNBDEC (GST_OBJECT_PARENT (pad));
GstBuffer *buf = gst_buffer_copy_on_write (in_data), *out;
guint8 *data = GST_BUFFER_DATA (buf);
gint size = GST_BUFFER_SIZE (buf), block, mode;
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf))
amrnbdec->ts = GST_BUFFER_TIMESTAMP (buf);
while (size >= 1) {
/* get size */
mode = (data[0] >> 3) & 0x0F;
block = block_size[mode] + 1;
if (size < block)
break;
/* get output */
out = gst_buffer_new_and_alloc (160 * 2);
GST_BUFFER_DURATION (out) = GST_SECOND * 160 /
(amrnbdec->rate * amrnbdec->channels);
GST_BUFFER_TIMESTAMP (out) = amrnbdec->ts;
amrnbdec->ts += GST_BUFFER_DURATION (out);
/* decode */
Decoder_Interface_Decode (amrnbdec->handle, data,
(short *) GST_BUFFER_DATA (out), 0);
data += block;
size -= block;
/* play */
gst_pad_push (amrnbdec->srcpad, GST_DATA (out));
}
gst_buffer_unref (buf);
}
static GstElementStateReturn
gst_amrnbdec_state_change (GstElement * element)
{
GstAmrnbDec *amrnbdec = GST_AMRNBDEC (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
if (!(amrnbdec->handle = Decoder_Interface_init ()))
return GST_STATE_FAILURE;
break;
case GST_STATE_PAUSED_TO_READY:
amrnbdec->ts = 0;
break;
case GST_STATE_READY_TO_NULL:
Decoder_Interface_exit (amrnbdec->handle);
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}