gstreamer/ext/faad/gstfaad.c
David Schleef 86db595f56 ext/faad/gstfaad.c: Fix negotiation.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
(gst_faad_chain): Fix negotiation.
* ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
key and button events.
* gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
dung heap of code.
* gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
depends on gconf
* gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
function to encourage better negotiation, particularly between
audioconvert and osssink.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):  Make some debugging
more important.
* gst/typefind/gsttypefindfunctions.c:  Fix mistake in flash
typefinding.
* gst/vbidec/vbiscreen.c:  Add glib header
* pkgconfig/gstreamer-play.pc.in:  Depends on gst-interfaces.
2004-03-06 00:42:20 +00:00

454 lines
12 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"
GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE (
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
"systemstream = (bool) FALSE, "
"mpegversion = { (int) 2, (int) 4 }"
)
);
GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE (
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (bool) TRUE, "
"width = (int) { 16, 24, 32 }, "
"depth = (int) { 16, 24, 32 }, "
"rate = (int) [ 8000, 96000 ], "
"channels = (int) [ 1, 6 ]; "
"audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"depth = (int) { 32, 64 }, "
"rate = (int) [ 8000, 96000 ], "
"channels = (int) [ 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,
const GstCaps *caps);
static GstPadLinkReturn
gst_faad_srcconnect (GstPad *pad,
const GstCaps *caps);
static GstCaps *gst_faad_srcgetcaps (GstPad *pad);
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_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_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_static_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_static_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);
gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);
}
static GstPadLinkReturn
gst_faad_sinkconnect (GstPad *pad,
const GstCaps *caps)
{
/* 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)
{
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_simple ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
NULL);
break;
case FAAD_FMT_24BIT:
caps = gst_caps_new_simple ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 24,
"depth", G_TYPE_INT, 24,
NULL);
break;
case FAAD_FMT_32BIT:
caps = gst_caps_new_simple ("audio/x-raw-int",
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 32,
"depth", G_TYPE_INT, 32,
NULL);
break;
case FAAD_FMT_FLOAT:
caps = gst_caps_new_simple ("audio/x-raw-float",
"depth", G_TYPE_INT, 32,
NULL);
break;
case FAAD_FMT_DOUBLE:
caps = gst_caps_new_simple ("audio/x-raw-float",
"depth", G_TYPE_INT, 64,
NULL);
break;
default:
caps = gst_caps_new_empty ();
break;
}
if (!gst_caps_is_empty (caps)) {
GstStructure *structure = gst_caps_get_structure (caps, 0);
if (faad->samplerate != -1) {
gst_structure_set (structure,
"rate", G_TYPE_INT, faad->samplerate,
NULL);
} else {
gst_structure_set (structure,
"rate", GST_TYPE_INT_RANGE, 8000, 96000,
NULL);
}
if (faad->channels != -1) {
gst_structure_set (structure,
"channels", G_TYPE_INT, faad->channels,
NULL);
} else {
gst_structure_set (structure,
"channels", GST_TYPE_INT_RANGE, 1, 6,
NULL);
}
gst_structure_set (structure,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
NULL);
}
return caps;
}
return gst_caps_copy (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
}
static GstPadLinkReturn
gst_faad_srcconnect (GstPad *pad,
const GstCaps *caps)
{
GstStructure *structure;
const gchar *mimetype;
gint fmt = 0;
gint depth;
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
if (!faad->handle ||
(faad->samplerate == -1 || faad->channels == -1)) {
return GST_PAD_LINK_DELAYED;
}
structure = gst_caps_get_structure (caps, 0);
mimetype = gst_structure_get_name (structure);
if (!strcmp (mimetype, "audio/x-raw-int")) {
gint width;
if (!gst_structure_get_int (structure, "depth", &depth) ||
!gst_structure_get_int (structure, "width", &width))
return GST_PAD_LINK_REFUSED;
if (depth != width)
return GST_PAD_LINK_REFUSED;
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_structure_get_int (structure, "depth", &depth))
return GST_PAD_LINK_REFUSED;
switch (depth) {
case 32:
fmt = FAAD_FMT_FLOAT;
break;
case 64:
fmt = FAAD_FMT_DOUBLE;
break;
}
}
if (fmt) {
GstCaps *newcaps, *intersect;
faacDecConfiguration *conf;
conf = faacDecGetCurrentConfiguration (faad->handle);
conf->outputFormat = fmt;
faacDecSetConfiguration (faad->handle, conf);
/* FIXME: handle return value, how? */
newcaps = gst_faad_srcgetcaps (pad);
g_assert (gst_caps_is_fixed (newcaps));
intersect = gst_caps_intersect (newcaps, caps);
gst_caps_free (newcaps);
if (!gst_caps_is_empty (intersect)) {
gst_caps_free (intersect);
faad->bps = depth / 8;
return GST_PAD_LINK_OK;
}
gst_caps_free (intersect);
}
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;
gst_pad_renegotiate (faad->srcpad);
#if 0
if (gst_faad_srcconnect (faad->srcpad,
gst_pad_get_allowed_caps (faad->srcpad)) <= 0) {
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
gst_buffer_unref (buf);
return;
}
#endif
}
out = faacDecDecode (faad->handle, &info,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
if (info.error) {
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("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;
gst_pad_renegotiate (faad->srcpad);
#if 0
if (gst_faad_srcconnect (faad->srcpad,
gst_pad_get_allowed_caps (faad->srcpad)) <= 0) {
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
gst_buffer_unref (buf);
return;
}
#endif
}
if (info.samples == 0) {
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_PACKAGE,
GST_ORIGIN
)