--- /dev/null
+/* 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_lame_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;
+
+ 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);
+ 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;
+ conf->defObjectType = LC;
+ conf->defSampleRate = faad->samplerate;
+ 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 * faad->channels / 8;
+ return GST_PAD_LINK_OK;
+ }
+ }
+ }
+
+ 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 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
+)
projects / platform / upstream / gstreamer.git / commitdiff