gstreamer/ext/faad/gstfaad.c
Ronald S. Bultje 8d529948e6 BPS is per-channel BPS, not total BPS... Ohwell
Original commit message from CVS:
BPS is per-channel BPS, not total BPS... Ohwell
2003-11-22 11:35:11 +00:00

478 lines
13 KiB
C

/* GStreamer FAAD (Free AAC Decoder) plugin
* Copyright (C) 2003 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 <string.h>
#include "gstfaad.h"
GST_PAD_TEMPLATE_FACTORY (sink_template,
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"faad_mpeg_templ",
"audio/mpeg",
"systemstream", GST_PROPS_BOOLEAN (FALSE),
"mpegversion", GST_PROPS_LIST (
GST_PROPS_INT (2),
GST_PROPS_INT (4)
)
)
);
GST_PAD_TEMPLATE_FACTORY (src_template,
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"faad_int_templ",
"audio/x-raw-int",
"endianness", GST_PROPS_INT (G_BYTE_ORDER),
"signed", GST_PROPS_BOOLEAN (TRUE),
"width", GST_PROPS_LIST (
GST_PROPS_INT (16),
GST_PROPS_INT (24),
GST_PROPS_INT (32)
),
"depth", GST_PROPS_LIST (
GST_PROPS_INT (16),
GST_PROPS_INT (24),
GST_PROPS_INT (32)
),
"rate", GST_PROPS_INT_RANGE (8000, 96000),
"channels", GST_PROPS_INT_RANGE (1, 6)
),
GST_CAPS_NEW (
"faad_float_templ",
"audio/x-raw-float",
"endianness", GST_PROPS_INT (G_BYTE_ORDER),
"depth", GST_PROPS_LIST (
GST_PROPS_INT (32), /* float */
GST_PROPS_INT (64) /* double */
),
"rate", GST_PROPS_INT_RANGE (8000, 96000),
"channels", GST_PROPS_INT_RANGE (1, 6)
)
);
static void gst_faad_base_init (GstFaadClass *klass);
static void gst_faad_class_init (GstFaadClass *klass);
static void gst_faad_init (GstFaad *faad);
static GstPadLinkReturn
gst_faad_sinkconnect (GstPad *pad,
GstCaps *caps);
static GstPadLinkReturn
gst_faad_srcconnect (GstPad *pad,
GstCaps *caps);
static GstCaps *gst_faad_srcgetcaps (GstPad *pad,
GstCaps *caps);
static void gst_faad_chain (GstPad *pad,
GstData *data);
static GstElementStateReturn
gst_faad_change_state (GstElement *element);
static GstElementClass *parent_class = NULL;
/* static guint gst_faad_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_faad_get_type (void)
{
static GType gst_faad_type = 0;
if (!gst_faad_type) {
static const GTypeInfo gst_faad_info = {
sizeof (GstFaadClass),
(GBaseInitFunc) gst_faad_base_init,
NULL,
(GClassInitFunc) gst_faad_class_init,
NULL,
NULL,
sizeof(GstFaad),
0,
(GInstanceInitFunc) gst_faad_init,
};
gst_faad_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstFaad",
&gst_faad_info, 0);
}
return gst_faad_type;
}
static void
gst_faad_base_init (GstFaadClass *klass)
{
GstElementDetails gst_faad_details = {
"Free AAC Decoder (FAAD)",
"Codec/Audio/Decoder",
"Free MPEG-2/4 AAC decoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>",
};
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
GST_PAD_TEMPLATE_GET (src_template));
gst_element_class_add_pad_template (element_class,
GST_PAD_TEMPLATE_GET (sink_template));
gst_element_class_set_details (element_class, &gst_faad_details);
}
static void
gst_faad_class_init (GstFaadClass *klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gstelement_class->change_state = gst_faad_change_state;
}
static void
gst_faad_init (GstFaad *faad)
{
faad->handle = NULL;
faad->samplerate = -1;
faad->channels = -1;
GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE);
faad->sinkpad = gst_pad_new_from_template (
GST_PAD_TEMPLATE_GET (sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (faad), faad->sinkpad);
gst_pad_set_chain_function (faad->sinkpad, gst_faad_chain);
gst_pad_set_link_function (faad->sinkpad, gst_faad_sinkconnect);
faad->srcpad = gst_pad_new_from_template (
GST_PAD_TEMPLATE_GET (src_template), "src");
gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad);
gst_pad_set_link_function (faad->srcpad, gst_faad_srcconnect);
/* This was originally intended as a getcaps() function, but
* in the end, we needed a srcconnect() function, so this is
* not really useful. However, srcconnect() uses it, so it is
* still there... */
/*gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);*/
}
static GstPadLinkReturn
gst_faad_sinkconnect (GstPad *pad,
GstCaps *caps)
{
if (!GST_CAPS_IS_FIXED (caps))
return GST_PAD_LINK_DELAYED;
/* oh, we really don't care what's in here. We'll
* get AAC audio (MPEG-2/4) anyway, so why bother? */
return GST_PAD_LINK_OK;
}
static GstCaps *
gst_faad_srcgetcaps (GstPad *pad,
GstCaps *caps)
{
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
if (faad->handle != NULL &&
faad->channels != -1 && faad->samplerate != -1) {
faacDecConfiguration *conf;
GstCaps *caps;
conf = faacDecGetCurrentConfiguration (faad->handle);
switch (conf->outputFormat) {
case FAAD_FMT_16BIT:
caps = GST_CAPS_NEW ("faad_src_int16",
"audio/x-raw-int",
"signed", GST_PROPS_BOOLEAN (TRUE),
"width", GST_PROPS_INT (16),
"depth", GST_PROPS_INT (16));
break;
case FAAD_FMT_24BIT:
caps = GST_CAPS_NEW ("faad_src_int24",
"audio/x-raw-int",
"signed", GST_PROPS_BOOLEAN (TRUE),
"width", GST_PROPS_INT (24),
"depth", GST_PROPS_INT (24));
break;
case FAAD_FMT_32BIT:
caps = GST_CAPS_NEW ("faad_src_int32",
"audio/x-raw-int",
"signed", GST_PROPS_BOOLEAN (TRUE),
"width", GST_PROPS_INT (32),
"depth", GST_PROPS_INT (32));
break;
case FAAD_FMT_FLOAT:
caps = GST_CAPS_NEW ("faad_src_float32",
"audio/x-raw-float",
"depth", GST_PROPS_INT (32));
break;
case FAAD_FMT_DOUBLE:
caps = GST_CAPS_NEW ("faad_src_float64",
"audio/x-raw-float",
"depth", GST_PROPS_INT (64));
break;
default:
caps = GST_CAPS_NONE;
break;
}
if (caps) {
GstPropsEntry *samplerate, *channels, *endianness;
if (faad->samplerate != -1) {
samplerate = gst_props_entry_new ("rate",
GST_PROPS_INT (faad->samplerate));
} else {
samplerate = gst_props_entry_new ("rate",
GST_PROPS_INT_RANGE (8000, 96000));
}
gst_props_add_entry (caps->properties, samplerate);
if (faad->channels != -1) {
channels = gst_props_entry_new ("channels",
GST_PROPS_INT (faad->channels));
} else {
channels = gst_props_entry_new ("channels",
GST_PROPS_INT_RANGE (1, 6));
}
gst_props_add_entry (caps->properties, channels);
endianness = gst_props_entry_new ("endianness",
GST_PROPS_INT (G_BYTE_ORDER));
gst_props_add_entry (caps->properties, endianness);
}
return caps;
}
return gst_pad_template_get_caps (
GST_PAD_TEMPLATE_GET (src_template));
}
static GstPadLinkReturn
gst_faad_srcconnect (GstPad *pad,
GstCaps *caps)
{
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
GstCaps *t;
if (!faad->handle ||
(faad->samplerate == -1 || faad->channels == -1)) {
return GST_PAD_LINK_DELAYED;
}
/* we do samplerate/channels ourselves */
for (t = caps; t != NULL; t = t->next) {
gst_props_remove_entry_by_name (t->properties, "rate");
gst_props_remove_entry_by_name (t->properties, "channels");
}
/* go through list */
caps = gst_caps_normalize (caps);
for ( ; caps != NULL; caps = caps->next) {
const gchar *mimetype = gst_caps_get_mime (caps);
gint depth = 0, fmt = 0;
if (!strcmp (mimetype, "audio/x-raw-int")) {
gint width = 0;
if (gst_caps_has_fixed_property (caps, "depth") &&
gst_caps_has_fixed_property (caps, "width"))
gst_caps_get (caps, "depth", &depth,
"width", &width, NULL);
if (depth != width)
continue;
switch (depth) {
case 16:
fmt = FAAD_FMT_16BIT;
break;
case 24:
fmt = FAAD_FMT_24BIT;
break;
case 32:
fmt = FAAD_FMT_32BIT;
break;
}
} else {
if (gst_caps_has_fixed_property (caps, "depth"))
gst_caps_get_int (caps, "depth", &depth);
switch (depth) {
case 32:
fmt = FAAD_FMT_FLOAT;
break;
case 64:
fmt = FAAD_FMT_DOUBLE;
break;
}
}
if (fmt) {
GstCaps *newcaps;
faacDecConfiguration *conf;
conf = faacDecGetCurrentConfiguration (faad->handle);
conf->outputFormat = fmt;
faacDecSetConfiguration (faad->handle, conf);
/* FIXME: handle return value, how? */
newcaps = gst_faad_srcgetcaps (pad, NULL);
g_assert (GST_CAPS_IS_FIXED (newcaps));
if (gst_pad_try_set_caps (pad, newcaps) > 0) {
faad->bps = depth / 8;
return GST_PAD_LINK_DONE;
}
}
}
return GST_PAD_LINK_REFUSED;
}
static void
gst_faad_chain (GstPad *pad,
GstData *data)
{
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
GstBuffer *buf, *outbuf;
faacDecFrameInfo info;
void *out;
if (GST_IS_EVENT (data)) {
GstEvent *event = GST_EVENT (data);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
gst_element_set_eos (GST_ELEMENT (faad));
gst_pad_push (faad->srcpad, data);
return;
default:
gst_pad_event_default (pad, event);
return;
}
}
buf = GST_BUFFER (data);
if (faad->samplerate == -1 || faad->channels == -1) {
gulong samplerate;
guchar channels;
faacDecInit (faad->handle,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
&samplerate, &channels);
faad->samplerate = samplerate;
faad->channels = channels;
if (gst_faad_srcconnect (faad->srcpad,
gst_pad_get_allowed_caps (faad->srcpad)) <= 0) {
gst_element_error (GST_ELEMENT (faad),
"Failed to negotiate output format with next element");
gst_buffer_unref (buf);
return;
}
}
out = faacDecDecode (faad->handle, &info,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
if (info.error) {
gst_element_error (GST_ELEMENT (faad),
"Failed to decode buffer: %s",
faacDecGetErrorMessage (info.error));
gst_buffer_unref (buf);
return;
}
if (info.samplerate != faad->samplerate ||
info.channels != faad->channels) {
faad->samplerate = info.samplerate;
faad->channels = info.channels;
if (gst_faad_srcconnect (faad->srcpad,
gst_pad_get_allowed_caps (faad->srcpad)) <= 0) {
gst_element_error (GST_ELEMENT (faad),
"Failed to re-negotiate format with next element");
gst_buffer_unref (buf);
return;
}
}
/* FIXME: did it handle the whole buffer? */
outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
/* ugh */
memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
gst_buffer_unref (buf);
gst_pad_push (faad->srcpad, GST_DATA (outbuf));
}
static GstElementStateReturn
gst_faad_change_state (GstElement *element)
{
GstFaad *faad = GST_FAAD (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
if (!(faad->handle = faacDecOpen ()))
return GST_STATE_FAILURE;
break;
case GST_STATE_PAUSED_TO_READY:
faad->samplerate = -1;
faad->channels = -1;
break;
case GST_STATE_READY_TO_NULL:
faacDecClose (faad->handle);
faad->handle = NULL;
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static gboolean
plugin_init (GstPlugin *plugin)
{
return gst_element_register (plugin, "faad",
GST_RANK_PRIMARY,
GST_TYPE_FAAD);
}
GST_PLUGIN_DEFINE (
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"faad",
"Free AAC Decoder (FAAD)",
plugin_init,
VERSION,
"GPL",
GST_COPYRIGHT,
GST_PACKAGE,
GST_ORIGIN
)