ext/wavpack/gstwavpackdec.c - imx-gst-plugins-good - Git at Google

 /* 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"

 GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
 #define GST_CAT_DEFAULT gst_wavpack_dec_debug

 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, 32 }, "
         "depth = (int) { 8, 16, 24, 32 }, "
         "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 GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
 static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);

 GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT)

      static gboolean gst_wavpack_dec_setcaps (GstPad * pad, GstCaps * caps)
 {
   GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
   GstStructure *structure;
   GstCaps *srccaps;
   gint bits, rate, channels;

   structure = gst_caps_get_structure (caps, 0);
   if (!gst_structure_get_int (structure, "rate", &rate) ||
       !gst_structure_get_int (structure, "channels", &channels) ||
       !gst_structure_get_int (structure, "width", &bits)) {
     return FALSE;
   }

   wavpackdec->samplerate = rate;
   wavpackdec->channels = channels;
   wavpackdec->width = bits;

 /* 32-bit output seems to be in fact 32 bit int (e.g. Prod_Girls.wv) */
 /*  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, G_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, G_LITTLE_ENDIAN, NULL);
   }
 */
 /*  gst_pad_set_caps (wavpackdec->sinkpad, caps); */

   gst_pad_set_caps (wavpackdec->srcpad, srccaps);
   gst_pad_use_fixed_caps (wavpackdec->srcpad);

   return TRUE;
 }

 #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

 static void
 gst_wavpack_dec_base_init (gpointer klass)
 {
   static const GstElementDetails plugin_details =
       GST_ELEMENT_DETAILS ("WavePack audio 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);
   wavpackdec->decodebuf = NULL;

   /* FIXME: what about wavpackdec->stream and wavpackdec->context? (tpm) */

   G_OBJECT_CLASS (parent_class)->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;

   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 gboolean
 gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
 {
   GstWavpackDec *dec;

   dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));

   GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_NEWSEGMENT:{
       /* TODO: save current segment so we can do clipping, for now
        * we'll just leave the clipping to the audio sink */
       break;
     }
     default:
       break;
   }

   gst_object_unref (dec);

   return gst_pad_event_default (pad, event);
 }

 static void
 gst_wavpack_dec_init (GstWavpackDec * wavpackdec, GstWavpackDecClass * gklass)
 {
   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_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
   gst_pad_set_setcaps_function (wavpackdec->sinkpad,
       GST_DEBUG_FUNCPTR (gst_wavpack_dec_setcaps));
   gst_pad_set_event_function (wavpackdec->sinkpad,
       GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));

 #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_DEBUG_FUNCPTR (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_stamp (outbuf, buf);

   gst_buffer_unref (buf);
   if (cbuf) {
     gst_buffer_unref (cbuf);
   }

   gst_buffer_set_caps (outbuf, GST_PAD_CAPS (wavpackdec->srcpad));

   GST_LOG_OBJECT (wavpackdec, "pushing buffer with time %" GST_TIME_FORMAT,
       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));

   ret = gst_pad_push (wavpackdec->srcpad, outbuf);
   if (ret != GST_FLOW_OK) {
     GST_DEBUG_OBJECT (wavpackdec, "pad_push: %s", gst_flow_get_name (ret));
   }

   return ret;
 }

 gboolean
 gst_wavpack_dec_plugin_init (GstPlugin * plugin)
 {
   if (!gst_element_register (plugin, "wavpackdec",
           GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
     return FALSE;

   GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
       "wavpack decoder");

   return TRUE;
 }
	/* 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"

	GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
	#define GST_CAT_DEFAULT gst_wavpack_dec_debug

	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, 32 }, "
	"depth = (int) { 8, 16, 24, 32 }, "
	"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 GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
	static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);

	GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT)

	static gboolean gst_wavpack_dec_setcaps (GstPad * pad, GstCaps * caps)
	{
	GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
	GstStructure *structure;
	GstCaps *srccaps;
	gint bits, rate, channels;

	structure = gst_caps_get_structure (caps, 0);
	if (!gst_structure_get_int (structure, "rate", &rate) \|\|
	!gst_structure_get_int (structure, "channels", &channels) \|\|
	!gst_structure_get_int (structure, "width", &bits)) {
	return FALSE;
	}

	wavpackdec->samplerate = rate;
	wavpackdec->channels = channels;
	wavpackdec->width = bits;

	/* 32-bit output seems to be in fact 32 bit int (e.g. Prod_Girls.wv) */
	/* 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, G_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, G_LITTLE_ENDIAN, NULL);
	}
	*/
	/* gst_pad_set_caps (wavpackdec->sinkpad, caps); */

	gst_pad_set_caps (wavpackdec->srcpad, srccaps);
	gst_pad_use_fixed_caps (wavpackdec->srcpad);

	return TRUE;
	}

	#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

	static void
	gst_wavpack_dec_base_init (gpointer klass)
	{
	static const GstElementDetails plugin_details =
	GST_ELEMENT_DETAILS ("WavePack audio 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);
	wavpackdec->decodebuf = NULL;

	/* FIXME: what about wavpackdec->stream and wavpackdec->context? (tpm) */

	G_OBJECT_CLASS (parent_class)->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;

	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 gboolean
	gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
	{
	GstWavpackDec *dec;

	dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));

	GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
	switch (GST_EVENT_TYPE (event)) {
	case GST_EVENT_NEWSEGMENT:{
	/* TODO: save current segment so we can do clipping, for now
	* we'll just leave the clipping to the audio sink */
	break;
	}
	default:
	break;
	}

	gst_object_unref (dec);

	return gst_pad_event_default (pad, event);
	}

	static void
	gst_wavpack_dec_init (GstWavpackDec * wavpackdec, GstWavpackDecClass * gklass)
	{
	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_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
	gst_pad_set_setcaps_function (wavpackdec->sinkpad,
	GST_DEBUG_FUNCPTR (gst_wavpack_dec_setcaps));
	gst_pad_set_event_function (wavpackdec->sinkpad,
	GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));

	#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_DEBUG_FUNCPTR (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_stamp (outbuf, buf);

	gst_buffer_unref (buf);
	if (cbuf) {
	gst_buffer_unref (cbuf);
	}

	gst_buffer_set_caps (outbuf, GST_PAD_CAPS (wavpackdec->srcpad));

	GST_LOG_OBJECT (wavpackdec, "pushing buffer with time %" GST_TIME_FORMAT,
	GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));

	ret = gst_pad_push (wavpackdec->srcpad, outbuf);
	if (ret != GST_FLOW_OK) {
	GST_DEBUG_OBJECT (wavpackdec, "pad_push: %s", gst_flow_get_name (ret));
	}

	return ret;
	}

	gboolean
	gst_wavpack_dec_plugin_init (GstPlugin * plugin)
	{
	if (!gst_element_register (plugin, "wavpackdec",
	GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
	return FALSE;

	GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
	"wavpack decoder");

	return TRUE;
	}