gstreamer/ext/wavpack/gstwavpackdec.c

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/* GStreamer Wavpack plugin
* (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
*
* gstwavpackdec.c: raw Wavpack bitstream decoder
*
* 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.
*/
#include <gst/gst.h>
#include <math.h>
#include <string.h>
#include <wavpack/wavpack.h>
#include "gstwavpackdec.h"
#include "gstwavpackcommon.h"
/* Filter signals and args */
enum
{
LAST_SIGNAL
};
enum
{
ARG_0
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) { 8, 16, 24, 32 }, "
"channels = (int) { 1, 2 }, "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
static GstStaticPadTemplate wvc_sink_factory =
GST_STATIC_PAD_TEMPLATE ("wvcsink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) { 8, 16, 24 }, "
"depth = (int) { 8, 16, 24 }, "
"channels = (int) { 1, 2 }, "
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) LITTLE_ENDIAN, "
"signed = (boolean) true;"
"audio/x-raw-float, "
"width = (int) 32, "
"channels = (int) { 1, 2 }, "
"rate = (int) [ 6000, 192000 ], " "endianness = (int) LITTLE_ENDIAN")
);
static void gst_wavpack_dec_class_init (GstWavpackDecClass * klass);
static void gst_wavpack_dec_base_init (GstWavpackDecClass * klass);
static void gst_wavpack_dec_init (GstWavpackDec * wavpackdec);
static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
static GstElementClass *parent = NULL;
static GstPadLinkReturn
gst_wavpack_dec_link (GstPad * pad, GstPad * peer)
{
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstStructure *structure;
GstCaps *srccaps;
gint bits;
if (!gst_caps_is_fixed (GST_PAD_CAPS (peer)))
return GST_PAD_LINK_REFUSED;
structure = gst_caps_get_structure (GST_PAD_CAPS (peer), 0);
gst_structure_get_int (structure, "rate",
(gint32 *) & wavpackdec->samplerate);
gst_structure_get_int (structure, "channels",
(gint *) & wavpackdec->channels);
gst_structure_get_int (structure, "width", &bits);
wavpackdec->width = bits;
if (bits != 32) {
srccaps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, wavpackdec->samplerate,
"channels", G_TYPE_INT, wavpackdec->channels,
"depth", G_TYPE_INT, bits,
"width", G_TYPE_INT, bits,
"endianness", G_TYPE_INT, LITTLE_ENDIAN,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
} else {
srccaps = gst_caps_new_simple ("audio/x-raw-float",
"rate", G_TYPE_INT, wavpackdec->samplerate,
"channels", G_TYPE_INT, wavpackdec->channels,
"width", G_TYPE_INT, 32, "endianness", G_TYPE_INT, LITTLE_ENDIAN, NULL);
}
gst_pad_set_caps (wavpackdec->srcpad, srccaps);
gst_pad_use_fixed_caps (wavpackdec->srcpad);
return GST_PAD_LINK_OK;
}
#if 0
static GstPadLinkReturn
gst_wavpack_dec_wvclink (GstPad * pad, GstPad * peer)
{
if (!gst_caps_is_fixed (GST_PAD_CAPS (peer)))
return GST_PAD_LINK_REFUSED;
return GST_PAD_LINK_OK;
}
#endif
GType
gst_wavpack_dec_get_type (void)
{
static GType plugin_type = 0;
if (!plugin_type) {
static const GTypeInfo plugin_info = {
sizeof (GstWavpackDecClass),
(GBaseInitFunc) gst_wavpack_dec_base_init,
NULL,
(GClassInitFunc) gst_wavpack_dec_class_init,
NULL,
NULL,
sizeof (GstWavpackDec),
0,
(GInstanceInitFunc) gst_wavpack_dec_init,
};
plugin_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstWavpackDec", &plugin_info, 0);
}
return plugin_type;
}
static void
gst_wavpack_dec_base_init (GstWavpackDecClass * klass)
{
static GstElementDetails plugin_details = {
"WAVPACK decoder",
"Codec/Decoder/Audio",
"Decode Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>"
};
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&wvc_sink_factory));
gst_element_class_set_details (element_class, &plugin_details);
}
static void
gst_wavpack_dec_dispose (GObject * object)
{
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (object);
g_free (wavpackdec->decodebuf);
G_OBJECT_CLASS (parent)->dispose (object);
}
static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent = g_type_class_ref (GST_TYPE_ELEMENT);
gobject_class->dispose = gst_wavpack_dec_dispose;
}
static gboolean
gst_wavpack_dec_src_query (GstPad * pad, GstQuery * query)
{
return gst_pad_query_default (pad, query);
}
static void
gst_wavpack_dec_init (GstWavpackDec * wavpackdec)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackdec);
wavpackdec->sinkpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"sink"), "sink");
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->sinkpad);
gst_pad_set_chain_function (wavpackdec->sinkpad, gst_wavpack_dec_chain);
gst_pad_set_link_function (wavpackdec->sinkpad, gst_wavpack_dec_link);
#if 0
wavpackdec->wvcsinkpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"wvcsink"), "wvcsink");
gst_pad_set_link_function (wavpackdec->wvcsinkpad, gst_wavpack_dec_wvclink);
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->wvcsinkpad);
#endif
wavpackdec->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"src"), "src");
gst_pad_use_fixed_caps (wavpackdec->srcpad);
gst_pad_set_query_function (wavpackdec->srcpad, gst_wavpack_dec_src_query);
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->srcpad);
wavpackdec->decodebuf = NULL;
wavpackdec->decodebuf_size = 0;
wavpackdec->stream = (WavpackStream *) g_malloc0 (sizeof (WavpackStream));
wavpackdec->context = (WavpackContext *) g_malloc0 (sizeof (WavpackContext));
}
static void
gst_wavpack_dec_setup_context (GstWavpackDec * wavpackdec, guchar * data,
guchar * cdata)
{
WavpackContext *context = wavpackdec->context;
WavpackStream *stream = wavpackdec->stream;
guint buffer_size;
memset (context, 0, sizeof (context));
context->open_flags = 0;
context->current_stream = 0;
context->num_streams = 1;
memset (stream, 0, sizeof (stream));
context->streams[0] = stream;
gst_wavpack_read_header (&stream->wphdr, data);
stream->blockbuff = data;
if (cdata) {
context->wvc_flag = TRUE;
gst_wavpack_read_header (&stream->wphdr, cdata);
stream->block2buff = cdata;
} else {
context->wvc_flag = FALSE;
}
buffer_size =
stream->wphdr.block_samples * wavpackdec->channels * sizeof (int32_t);
if (wavpackdec->decodebuf_size < buffer_size) {
wavpackdec->decodebuf =
(int32_t *) g_realloc (wavpackdec->decodebuf, buffer_size);
wavpackdec->decodebuf_size = buffer_size;
}
unpack_init (context);
}
static GstBuffer *
gst_wavpack_dec_format_samples (GstWavpackDec * wavpackdec, int32_t * samples,
guint num_samples)
{
GstBuffer *buf;
gint i;
guint8 *dst;
int32_t temp;
buf =
gst_buffer_new_and_alloc (num_samples * wavpackdec->width / 8 *
wavpackdec->channels);
dst = (guint8 *) GST_BUFFER_DATA (buf);
switch (wavpackdec->width) {
case 8:
for (i = 0; i < num_samples * wavpackdec->channels; ++i)
*dst++ = *samples++ + 128;
break;
case 16:
for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
*dst++ = (guint8) (temp = *samples++);
*dst++ = (guint8) (temp >> 8);
}
break;
case 24:
for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
*dst++ = (guint8) (temp = *samples++);
*dst++ = (guint8) (temp >> 8);
*dst++ = (guint8) (temp >> 16);
}
break;
case 32:
for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
*dst++ = (guint8) (temp = *samples++);
*dst++ = (guint8) (temp >> 8);
*dst++ = (guint8) (temp >> 16);
*dst++ = (guint8) (temp >> 24);
}
break;
default:
break;
}
return buf;
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstBuffer *outbuf, *cbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
#if 0
if (gst_pad_is_linked (wavpackdec->wvcsinkpad)) {
if (GST_FLOW_OK != gst_pad_pull_range (wavpackdec->wvcsinkpad,
wavpackdec->wvcflushed_bytes, -1, &cbuf)) {
cbuf = NULL;
} else {
wavpackdec->wvcflushed_bytes += GST_BUFFER_SIZE (cbuf);
}
}
#endif
gst_wavpack_dec_setup_context (wavpackdec, GST_BUFFER_DATA (buf),
cbuf ? GST_BUFFER_DATA (cbuf) : NULL);
unpack_samples (wavpackdec->context, wavpackdec->decodebuf,
wavpackdec->context->streams[0]->wphdr.block_samples);
outbuf =
gst_wavpack_dec_format_samples (wavpackdec, wavpackdec->decodebuf,
wavpackdec->context->streams[0]->wphdr.block_samples);
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
gst_buffer_unref (buf);
if (cbuf) {
gst_buffer_unref (cbuf);
}
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (wavpackdec->srcpad));
if (GST_FLOW_OK != (ret = gst_pad_push (wavpackdec->srcpad, outbuf))) {
gst_buffer_unref (outbuf);
}
return ret;
}
gboolean
gst_wavpack_dec_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "wavpackdec",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC);
}