Initial import of libwavpack

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6056 a1c6a512-1295-4272-9138-f99709370657

12 files changed, 2190 insertions, 0 deletions

diff --git a/apps/codecs/libwavpack/LICENSE b/apps/codecs/libwavpack/LICENSE
new file mode 100644
index 0000000..3ee4858
--- /dev/null
+++ b/apps/codecs/libwavpack/LICENSE
@@ -0,0 +1,25 @@
+               Copyright (c) 1998 - 2004 Conifer Software
+                          All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are met:
+
+    * Redistributions of source code must retain the above copyright notice,
+      this list of conditions and the following disclaimer.
+    * Redistributions in binary form must reproduce the above copyright notice,
+      this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+    * Neither the name of Conifer Software nor the names of its contributors
+      may be used to endorse or promote products derived from this software
+      without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE FOR
+ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/apps/codecs/libwavpack/README b/apps/codecs/libwavpack/README
new file mode 100644
index 0000000..b373235
--- /dev/null
+++ b/apps/codecs/libwavpack/README
@@ -0,0 +1,52 @@
+////////////////////////////////////////////////////////////////////////////
+//                           **** WAVPACK ****                            //
+//                  Hybrid Lossless Wavefile Compressor                   //
+//              Copyright (c) 1998 - 2004 Conifer Software.               //
+//                          All Rights Reserved.                          //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+This package contains a tiny version of the WavPack 4.0 decoder that might
+be used in a "resource limited" CPU environment or form the basis for a
+hardware decoding implementation. It is packaged with a demo command-line
+program that accepts a WavPack audio file on stdin and outputs a RIFF wav
+file to stdout. The program is standard C, and a win32 executable is
+included which was compiled under MS Visual C++ 6.0 using this command:
+
+cl /O1 /DWIN32 wvfilter.c wputils.c unpack.c float.c metadata.c words.c bits.c
+
+WavPack data is read with a stream reading callback. No direct seeking is
+provided for, but it is possible to start decoding anywhere in a WavPack
+stream. In this case, WavPack will be able to provide the sample-accurate
+position when it synchs with the data and begins decoding.
+
+For demonstration purposes this uses a single static copy of the
+WavpackContext structure, so obviously it cannot be used for more than one
+file at a time. Also, this decoder will not handle "correction" files, plays
+only the first two channels of multi-channel files, and is limited in
+resolution in some large integer or floating point files (but always
+provides at least 24 bits of resolution). It also will not accept WavPack
+files from before version 4.0.
+
+To make this code viable on the greatest number of hardware platforms, the
+following are true:
+
+   speed is about 4x realtime on an AMD K6 300 MHz
+      ("high" mode 16/44 stereo; normal mode is about twice that fast)
+
+   no floating-point math required; just 32b * 32b = 32b int multiply
+
+   large data areas are static and less than 4K total
+   executable code and tables are less than 32K
+   no malloc / free usage
+
+To maintain compatibility on various platforms, the following conventions
+are used:
+
+   a "short" must be 16-bits
+   a "long" must be 32-bits
+   an "int" must be at least 16-bits, but may be larger
+   a "char" must default to signed
+
+
+Questions or comments should be directed to david@wavpack.com
diff --git a/apps/codecs/libwavpack/README.rockbox b/apps/codecs/libwavpack/README.rockbox
new file mode 100644
index 0000000..a74b0c5
--- /dev/null
+++ b/apps/codecs/libwavpack/README.rockbox
@@ -0,0 +1,15 @@
+Library: wavpack - Release 4.1 - September 14, 2004
+Imported: 2005-02-25 by Christian Gmeiner
+
+
+This directory contains a "tiny" decoder version of wavpack for version 4.x.
+
+LICENSING INFORMATION
+
+wavpack is released under the BSD License as described
+in the LICENSE file in this directory.
+
+
+IMPORT DETAILS
+
+Excluded is wvfilter.c, because it is only a test programm.
diff --git a/apps/codecs/libwavpack/SOURCES b/apps/codecs/libwavpack/SOURCES
new file mode 100644
index 0000000..9b79d70
--- /dev/null
+++ b/apps/codecs/libwavpack/SOURCES
@@ -0,0 +1,6 @@
+bits.c
+float.c
+metadata.c
+unpack.c
+words.c
+wputils.c
\ No newline at end of file
diff --git a/apps/codecs/libwavpack/bits.c b/apps/codecs/libwavpack/bits.c
new file mode 100644
index 0000000..25a2227
--- /dev/null
+++ b/apps/codecs/libwavpack/bits.c
@@ -0,0 +1,140 @@
+////////////////////////////////////////////////////////////////////////////
+//                           **** WAVPACK ****                            //
+//                  Hybrid Lossless Wavefile Compressor                   //
+//              Copyright (c) 1998 - 2004 Conifer Software.               //
+//                          All Rights Reserved.                          //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+// bits.c
+
+// This module provides utilities to support the BitStream structure which is
+// used to read and write all WavPack audio data streams. It also contains a
+// wrapper for the stream I/O functions and a set of functions dealing with
+// endian-ness, both for enhancing portability. Finally, a debug wrapper for
+// the malloc() system is provided.
+
+#include "wavpack.h"
+
+#include <string.h>
+#include <ctype.h>
+
+////////////////////////// Bitstream functions ////////////////////////////////
+
+// Open the specified BitStream and associate with the specified buffer.
+
+static void bs_read (Bitstream *bs);
+
+void bs_open_read (Bitstream *bs, uchar *buffer_start, uchar *buffer_end, read_stream file, ulong file_bytes)
+{
+    CLEAR (*bs);
+    bs->buf = buffer_start;
+    bs->end = buffer_end;
+
+    if (file) {
+	bs->ptr = bs->end - 1;
+	bs->file_bytes = file_bytes;
+	bs->file = file;
+    }
+    else
+	bs->ptr = bs->buf - 1;
+
+    bs->wrap = bs_read;
+}
+
+// This function is only called from the getbit() and getbits() macros when
+// the BitStream has been exhausted and more data is required. Sinve these
+// bistreams no longer access files, this function simple sets an error and
+// resets the buffer.
+
+static void bs_read (Bitstream *bs)
+{
+    if (bs->file && bs->file_bytes) {
+	ulong bytes_read, bytes_to_read = bs->end - bs->buf;
+
+	if (bytes_to_read > bs->file_bytes)
+	    bytes_to_read = bs->file_bytes;
+
+	bytes_read = bs->file (bs->buf, bytes_to_read);
+
+	if (bytes_read) {
+	    bs->end = bs->buf + bytes_read;
+	    bs->file_bytes -= bytes_read;
+	}
+	else {
+	    memset (bs->buf, -1, bs->end - bs->buf);
+	    bs->error = 1;
+	}
+    }
+    else
+	bs->error = 1;
+
+    if (bs->error)
+	memset (bs->buf, -1, bs->end - bs->buf);
+
+    bs->ptr = bs->buf;
+}
+
+/////////////////////// Endian Correction Routines ////////////////////////////
+
+void little_endian_to_native (void *data, char *format)
+{
+    uchar *cp = (uchar *) data;
+    long temp;
+
+    while (*format) {
+	switch (*format) {
+	    case 'L':
+		temp = cp [0] + ((long) cp [1] << 8) + ((long) cp [2] << 16) + ((long) cp [3] << 24);
+		* (long *) cp = temp;
+		cp += 4;
+		break;
+
+	    case 'S':
+		temp = cp [0] + (cp [1] << 8);
+		* (short *) cp = (short) temp;
+		cp += 2;
+		break;
+
+	    default:
+		if (isdigit (*format))
+		    cp += *format - '0';
+
+		break;
+	}
+
+	format++;
+    }
+}
+
+void native_to_little_endian (void *data, char *format)
+{
+    uchar *cp = (uchar *) data;
+    long temp;
+
+    while (*format) {
+	switch (*format) {
+	    case 'L':
+		temp = * (long *) cp;
+		*cp++ = (uchar) temp;
+		*cp++ = (uchar) (temp >> 8);
+		*cp++ = (uchar) (temp >> 16);
+		*cp++ = (uchar) (temp >> 24);
+		break;
+
+	    case 'S':
+		temp = * (short *) cp;
+		*cp++ = (uchar) temp;
+		*cp++ = (uchar) (temp >> 8);
+		break;
+
+	    default:
+		if (isdigit (*format))
+		    cp += *format - '0';
+
+		break;
+	}
+
+	format++;
+    }
+}
diff --git a/apps/codecs/libwavpack/float.c b/apps/codecs/libwavpack/float.c
new file mode 100644
index 0000000..3e678e8
--- /dev/null
+++ b/apps/codecs/libwavpack/float.c
@@ -0,0 +1,83 @@
+////////////////////////////////////////////////////////////////////////////
+//			     **** WAVPACK ****				  //
+//		    Hybrid Lossless Wavefile Compressor			  //
+//		Copyright (c) 1998 - 2004 Conifer Software.		  //
+//			    All Rights Reserved.			  //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+// float.c
+
+#include "wavpack.h"
+
+int read_float_info (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+    int bytecnt = wpmd->byte_length;
+    char *byteptr = wpmd->data;
+
+    if (bytecnt != 4)
+	return FALSE;
+
+    wps->float_flags = *byteptr++;
+    wps->float_shift = *byteptr++;
+    wps->float_max_exp = *byteptr++;
+    wps->float_norm_exp = *byteptr;
+    return TRUE;
+}
+
+void float_values (WavpackStream *wps, long *values, long num_values)
+{
+    while (num_values--) {
+	int shift_count = 0, exp = wps->float_max_exp;
+	f32 outval = { 0, 0, 0 };
+
+	if (*values) {
+	    *values <<= wps->float_shift;
+
+	    if (*values < 0) {
+		*values = -*values;
+		outval.sign = 1;
+	    }
+
+	    if (*values == 0x1000000)
+		outval.exponent = 255;
+	    else {
+		if (exp)
+		    while (!(*values & 0x800000) && --exp) {
+			shift_count++;
+			*values <<= 1;
+		    }
+
+		if (shift_count && (wps->float_flags & FLOAT_SHIFT_ONES))
+		    *values |= ((1 << shift_count) - 1);
+
+		outval.mantissa = *values;
+		outval.exponent = exp;
+	    }
+	}
+
+	* (f32 *) values++ = outval;
+    }
+}
+
+void float_normalize (long *values, long num_values, int delta_exp)
+{
+    f32 *fvalues = (f32 *) values, fzero = { 0, 0, 0 };
+    int exp;
+
+    if (!delta_exp)
+	return;
+
+    while (num_values--) {
+	if ((exp = fvalues->exponent) == 0 || exp + delta_exp <= 0)
+	    *fvalues = fzero;
+	else if (exp == 255 || (exp += delta_exp) >= 255) {
+	    fvalues->exponent = 255;
+	    fvalues->mantissa = 0;
+	}
+	else
+	    fvalues->exponent = exp;
+
+	fvalues++;
+    }
+}
diff --git a/apps/codecs/libwavpack/make.bat b/apps/codecs/libwavpack/make.bat
new file mode 100644
index 0000000..0ca72d7
--- /dev/null
+++ b/apps/codecs/libwavpack/make.bat
@@ -0,0 +1 @@
+cl /O1 /DWIN32 wvfilter.c wputils.c unpack.c float.c metadata.c words.c bits.c
diff --git a/apps/codecs/libwavpack/metadata.c b/apps/codecs/libwavpack/metadata.c
new file mode 100644
index 0000000..40ede99
--- /dev/null
+++ b/apps/codecs/libwavpack/metadata.c
@@ -0,0 +1,105 @@
+////////////////////////////////////////////////////////////////////////////
+//                           **** WAVPACK ****                            //
+//                  Hybrid Lossless Wavefile Compressor                   //
+//              Copyright (c) 1998 - 2003 Conifer Software.               //
+//                          All Rights Reserved.                          //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+// metadata.c
+
+// This module handles the metadata structure introduced in WavPack 4.0
+
+#include "wavpack.h"
+
+int read_metadata_buff (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+    uchar tchar;
+
+    if (!wpc->infile (&wpmd->id, 1) || !wpc->infile (&tchar, 1))
+	return FALSE;
+
+    wpmd->byte_length = tchar << 1;
+
+    if (wpmd->id & ID_LARGE) {
+	wpmd->id &= ~ID_LARGE;
+
+	if (!wpc->infile (&tchar, 1))
+	    return FALSE;
+
+	wpmd->byte_length += (long) tchar << 9; 
+
+	if (!wpc->infile (&tchar, 1))
+	    return FALSE;
+
+	wpmd->byte_length += (long) tchar << 17;
+    }
+
+    if (wpmd->id & ID_ODD_SIZE) {
+	wpmd->id &= ~ID_ODD_SIZE;
+	wpmd->byte_length--;
+    }
+
+    if (wpmd->byte_length && wpmd->byte_length <= sizeof (wpc->read_buffer)) {
+	ulong bytes_to_read = wpmd->byte_length + (wpmd->byte_length & 1);
+
+	if (wpc->infile (wpc->read_buffer, bytes_to_read) != (long) bytes_to_read) {
+	    wpmd->data = NULL;
+	    return FALSE;
+	}
+
+	wpmd->data = wpc->read_buffer;
+    }
+    else
+	wpmd->data = NULL;
+
+    return TRUE;
+}
+
+int process_metadata (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+    WavpackStream *wps = &wpc->stream;
+
+    switch (wpmd->id) {
+	case ID_DUMMY:
+	    return TRUE;
+
+	case ID_DECORR_TERMS:
+	    return read_decorr_terms (wps, wpmd);
+
+	case ID_DECORR_WEIGHTS:
+	    return read_decorr_weights (wps, wpmd);
+
+	case ID_DECORR_SAMPLES:
+	    return read_decorr_samples (wps, wpmd);
+
+	case ID_ENTROPY_VARS:
+	    return read_entropy_vars (wps, wpmd);
+
+	case ID_HYBRID_PROFILE:
+	    return read_hybrid_profile (wps, wpmd);
+
+	case ID_FLOAT_INFO:
+	    return read_float_info (wps, wpmd);
+
+	case ID_INT32_INFO:
+	    return read_int32_info (wps, wpmd);
+
+	case ID_CHANNEL_INFO:
+	    return read_channel_info (wpc, wpmd);
+
+	case ID_CONFIG_BLOCK:
+	    return read_config_info (wpc, wpmd);
+
+	case ID_WV_BITSTREAM:
+	    return init_wv_bitstream (wpc, wpmd);
+
+	case ID_SHAPING_WEIGHTS:
+	case ID_WVC_BITSTREAM:
+	case ID_WVX_BITSTREAM:
+	    return TRUE;
+
+	default:
+	    return (wpmd->id & ID_OPTIONAL_DATA) ? TRUE : FALSE;
+    }
+}
diff --git a/apps/codecs/libwavpack/unpack.c b/apps/codecs/libwavpack/unpack.c
new file mode 100644
index 0000000..e2e27b4
--- /dev/null
+++ b/apps/codecs/libwavpack/unpack.c
@@ -0,0 +1,576 @@
+////////////////////////////////////////////////////////////////////////////
+//			     **** WAVPACK ****				  //
+//		    Hybrid Lossless Wavefile Compressor			  //
+//		Copyright (c) 1998 - 2004 Conifer Software.		  //
+//			    All Rights Reserved.			  //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+// unpack.c
+
+// This module actually handles the decompression of the audio data, except
+// for the entropy decoding which is handled by the words.c module. For
+// maximum efficiency, the conversion is isolated to tight loops that handle
+// an entire buffer.
+
+#include "wavpack.h"
+
+#include <string.h>
+#include <math.h>
+
+#define LOSSY_MUTE
+
+//////////////////////////////// local macros /////////////////////////////////
+
+#define apply_weight_i(weight, sample) ((weight * sample + 512) >> 10)
+
+#define apply_weight_f(weight, sample) (((((sample & 0xffff) * weight) >> 9) + \
+    (((sample & ~0xffff) >> 9) * weight) + 1) >> 1)
+
+#define apply_weight(weight, sample) (sample != (short) sample ? \
+    apply_weight_f (weight, sample) : apply_weight_i (weight, sample))
+
+#define update_weight(weight, delta, source, result) \
+    if (source && result) weight -= ((((source ^ result) >> 30) & 2) - 1) * delta;
+
+#define update_weight_clip(weight, delta, source, result) \
+    if (source && result && ((source ^ result) < 0 ? (weight -= delta) < -1024 : (weight += delta) > 1024)) \
+	weight = weight < 0 ? -1024 : 1024;
+
+///////////////////////////// executable code ////////////////////////////////
+
+// This function initializes everything required to unpack a WavPack block
+// and must be called before unpack_samples() is called to obtain audio data.
+// It is assumed that the WavpackHeader has been read into the wps->wphdr
+// (in the current WavpackStream). This is where all the metadata blocks are
+// scanned up to the one containing the audio bitstream.
+
+int unpack_init (WavpackContext *wpc)
+{
+    WavpackStream *wps = &wpc->stream;
+    WavpackMetadata wpmd;
+
+    if (wps->wphdr.block_samples && wps->wphdr.block_index != (ulong) -1)
+	wps->sample_index = wps->wphdr.block_index;
+
+    wps->mute_error = FALSE;
+    wps->crc = 0xffffffff;
+    CLEAR (wps->wvbits);
+    CLEAR (wps->decorr_passes);
+    CLEAR (wps->w);
+
+    while (read_metadata_buff (wpc, &wpmd)) {
+	if (!process_metadata (wpc, &wpmd)) {
+	    strcpy (wpc->error_message, "invalid metadata!");
+	    return FALSE;
+	}
+
+	if (wpmd.id == ID_WV_BITSTREAM)
+	    break;
+    }
+
+    if (wps->wphdr.block_samples && !bs_is_open (&wps->wvbits)) {
+	strcpy (wpc->error_message, "invalid WavPack file!");
+	return FALSE;
+    }
+
+    if (wps->wphdr.block_samples) {
+	if ((wps->wphdr.flags & INT32_DATA) && wps->int32_sent_bits)
+	    wpc->lossy_blocks = TRUE;
+
+	if ((wps->wphdr.flags & FLOAT_DATA) &&
+	    wps->float_flags & (FLOAT_EXCEPTIONS | FLOAT_ZEROS_SENT | FLOAT_SHIFT_SENT | FLOAT_SHIFT_SAME))
+		wpc->lossy_blocks = TRUE;
+    }
+
+    return TRUE;
+}
+
+// This function initialzes the main bitstream for audio samples, which must
+// be in the "wv" file.
+
+int init_wv_bitstream (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+    WavpackStream *wps = &wpc->stream;
+
+    if (wpmd->data)
+	bs_open_read (&wps->wvbits, wpmd->data, (char *) wpmd->data + wpmd->byte_length, NULL, 0);
+    else if (wpmd->byte_length)
+	bs_open_read (&wps->wvbits, wpc->read_buffer, wpc->read_buffer + sizeof (wpc->read_buffer),
+	    wpc->infile, wpmd->byte_length + (wpmd->byte_length & 1));
+
+    return TRUE;
+}
+
+// Read decorrelation terms from specified metadata block into the
+// decorr_passes array. The terms range from -3 to 8, plus 17 & 18;
+// other values are reserved and generate errors for now. The delta
+// ranges from 0 to 7 with all values valid. Note that the terms are
+// stored in the opposite order in the decorr_passes array compared
+// to packing.
+
+int read_decorr_terms (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+    int termcnt = wpmd->byte_length;
+    uchar *byteptr = wpmd->data;
+    struct decorr_pass *dpp;
+
+    if (termcnt > MAX_NTERMS)
+	return FALSE;
+
+    wps->num_terms = termcnt;
+
+    for (dpp = wps->decorr_passes + termcnt - 1; termcnt--; dpp--) {
+	dpp->term = (int)(*byteptr & 0x1f) - 5;
+	dpp->delta = (*byteptr++ >> 5) & 0x7;
+
+	if (!dpp->term || dpp->term < -3 || (dpp->term > MAX_TERM && dpp->term < 17) || dpp->term > 18)
+	    return FALSE;
+    }
+
+    return TRUE;
+}
+
+// Read decorrelation weights from specified metadata block into the
+// decorr_passes array. The weights range +/-1024, but are rounded and
+// truncated to fit in signed chars for metadata storage. Weights are
+// separate for the two channels and are specified from the "last" term
+// (first during encode). Unspecified weights are set to zero.
+
+int read_decorr_weights (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+    int termcnt = wpmd->byte_length, tcount;
+    char *byteptr = wpmd->data;
+    struct decorr_pass *dpp;
+
+    if (!(wps->wphdr.flags & MONO_FLAG))
+	termcnt /= 2;
+
+    if (termcnt > wps->num_terms)
+	return FALSE;
+
+    for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
+	dpp->weight_A = dpp->weight_B = 0;
+
+    while (--dpp >= wps->decorr_passes && termcnt--) {
+	dpp->weight_A = restore_weight (*byteptr++);
+
+	if (!(wps->wphdr.flags & MONO_FLAG))
+	    dpp->weight_B = restore_weight (*byteptr++);
+    }
+
+    return TRUE;
+}
+
+// Read decorrelation samples from specified metadata block into the
+// decorr_passes array. The samples are signed 32-bit values, but are
+// converted to signed log2 values for storage in metadata. Values are
+// stored for both channels and are specified from the "last" term
+// (first during encode) with unspecified samples set to zero. The
+// number of samples stored varies with the actual term value, so
+// those must obviously come first in the metadata.
+
+int read_decorr_samples (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+    uchar *byteptr = wpmd->data;
+    uchar *endptr = byteptr + wpmd->byte_length;
+    struct decorr_pass *dpp;
+    int tcount;
+
+    for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
+	CLEAR (dpp->samples_A);
+	CLEAR (dpp->samples_B);
+    }
+
+    if (wps->wphdr.version == 0x402 && (wps->wphdr.flags & HYBRID_FLAG)) {
+	byteptr += 2;
+
+	if (!(wps->wphdr.flags & MONO_FLAG))
+	    byteptr += 2;
+    }
+
+    while (dpp-- > wps->decorr_passes && byteptr < endptr)
+	if (dpp->term > MAX_TERM) {
+	    dpp->samples_A [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+	    dpp->samples_A [1] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
+	    byteptr += 4;
+
+	    if (!(wps->wphdr.flags & MONO_FLAG)) {
+		dpp->samples_B [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+		dpp->samples_B [1] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
+		byteptr += 4;
+	    }
+	}
+	else if (dpp->term < 0) {
+	    dpp->samples_A [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+	    dpp->samples_B [0] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
+	    byteptr += 4;
+	}
+	else {
+	    int m = 0, cnt = dpp->term;
+
+	    while (cnt--) {
+		dpp->samples_A [m] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+		byteptr += 2;
+
+		if (!(wps->wphdr.flags & MONO_FLAG)) {
+		    dpp->samples_B [m] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+		    byteptr += 2;
+		}
+
+		m++;
+	    }
+	}
+
+    return byteptr == endptr;
+}
+
+// Read the int32 data from the specified metadata into the specified stream.
+// This data is used for integer data that has more than 24 bits of magnitude
+// or, in some cases, used to eliminate redundant bits from any audio stream.
+
+int read_int32_info (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+    int bytecnt = wpmd->byte_length;
+    char *byteptr = wpmd->data;
+
+    if (bytecnt != 4)
+	return FALSE;
+
+    wps->int32_sent_bits = *byteptr++;
+    wps->int32_zeros = *byteptr++;
+    wps->int32_ones = *byteptr++;
+    wps->int32_dups = *byteptr;
+    return TRUE;
+}
+
+// Read multichannel information from metadata. The first byte is the total
+// number of channels and the following bytes represent the channel_mask
+// as described for Microsoft WAVEFORMATEX.
+
+int read_channel_info (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+    int bytecnt = wpmd->byte_length, shift = 0;
+    char *byteptr = wpmd->data;
+    ulong mask = 0;
+
+    if (!bytecnt || bytecnt > 5)
+	return FALSE;
+
+    wpc->config.num_channels = *byteptr++;
+
+    while (--bytecnt) {
+	mask |= (ulong) *byteptr++ << shift;
+	shift += 8;
+    }
+
+    wpc->config.channel_mask = mask;
+    return TRUE;
+}
+
+// Read configuration information from metadata.
+
+int read_config_info (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+    int bytecnt = wpmd->byte_length;
+    uchar *byteptr = wpmd->data;
+
+    if (bytecnt >= 3) {
+	wpc->config.flags &= 0xff;
+	wpc->config.flags |= (long) *byteptr++ << 8;
+	wpc->config.flags |= (long) *byteptr++ << 16;
+	wpc->config.flags |= (long) *byteptr << 24;
+    }
+
+    return TRUE;
+}
+
+// This monster actually unpacks the WavPack bitstream(s) into the specified
+// buffer as 32-bit integers or floats (depending on orignal data). Lossy
+// samples will be clipped to their original limits (i.e. 8-bit samples are
+// clipped to -128/+127) but are still returned in longs. It is up to the
+// caller to potentially reformat this for the final output including any
+// multichannel distribution, block alignment or endian compensation. The
+// function unpack_init() must have been called and the entire WavPack block
+// must still be visible (although wps->blockbuff will not be accessed again).
+// For maximum clarity, the function is broken up into segments that handle
+// various modes. This makes for a few extra infrequent flag checks, but
+// makes the code easier to follow because the nesting does not become so
+// deep. For maximum efficiency, the conversion is isolated to tight loops
+// that handle an entire buffer. The function returns the total number of
+// samples unpacked, which can be less than the number requested if an error
+// occurs or the end of the block is reached.
+
+static void fixup_samples (WavpackStream *wps, long *buffer, ulong sample_count);
+
+long unpack_samples (WavpackContext *wpc, long *buffer, ulong sample_count)
+{
+    WavpackStream *wps = &wpc->stream;
+    ulong flags = wps->wphdr.flags, crc = wps->crc, i;
+    long mute_limit = (1L << ((flags & MAG_MASK) >> MAG_LSB)) + 2;
+    struct decorr_pass *dpp;
+    long read_word, *bptr;
+    int tcount, m = 0;
+
+    if (wps->sample_index + sample_count > wps->wphdr.block_index + wps->wphdr.block_samples)
+	sample_count = wps->wphdr.block_index + wps->wphdr.block_samples - wps->sample_index;
+
+    if (wps->mute_error) {
+	memset (buffer, 0, sample_count * (flags & MONO_FLAG ? 4 : 8));
+	wps->sample_index += sample_count;
+	return sample_count;
+    }
+
+    if (flags & HYBRID_FLAG)
+	mute_limit *= 2;
+
+    ///////////////////// handle version 4 mono data /////////////////////////
+
+    if (flags & MONO_FLAG)
+	for (bptr = buffer, i = 0; i < sample_count; ++i) {
+	    if ((read_word = get_word (wps, 0)) == WORD_EOF)
+		break;
+
+	    for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
+		long sam, temp;
+		int k;
+
+		if (dpp->term > MAX_TERM) {
+		    if (dpp->term & 1)
+			sam = 2 * dpp->samples_A [0] - dpp->samples_A [1];
+		    else
+			sam = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
+
+		    dpp->samples_A [1] = dpp->samples_A [0];
+		    k = 0;
+		}
+		else {
+		    sam = dpp->samples_A [m];
+		    k = (m + dpp->term) & (MAX_TERM - 1);
+		}
+
+		temp = apply_weight (dpp->weight_A, sam) + read_word;
+		update_weight (dpp->weight_A, dpp->delta, sam, read_word);
+		dpp->samples_A [k] = read_word = temp;
+	    }
+
+	    if (labs (read_word) > mute_limit)
+		break;
+
+	    m = (m + 1) & (MAX_TERM - 1);
+	    crc = crc * 3 + read_word;
+	    *bptr++ = read_word;
+	}
+
+    //////////////////// handle version 4 stereo data ////////////////////////
+
+    else
+	for (bptr = buffer, i = 0; i < sample_count; ++i) {
+	    long left, right, left2, right2;
+
+	    if ((left = get_word (wps, 0)) == WORD_EOF ||
+		(right = get_word (wps, 1)) == WORD_EOF)
+		    break;
+
+	    for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
+		if (dpp->term > 0) {
+		    long sam_A, sam_B;
+		    int k;
+
+		    if (dpp->term > MAX_TERM) {
+			if (dpp->term & 1) {
+			    sam_A = 2 * dpp->samples_A [0] - dpp->samples_A [1];
+			    sam_B = 2 * dpp->samples_B [0] - dpp->samples_B [1];
+			}
+			else {
+			    sam_A = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
+			    sam_B = (3 * dpp->samples_B [0] - dpp->samples_B [1]) >> 1;
+			}
+
+			dpp->samples_A [1] = dpp->samples_A [0];
+			dpp->samples_B [1] = dpp->samples_B [0];
+			k = 0;
+		    }
+		    else {
+			sam_A = dpp->samples_A [m];
+			sam_B = dpp->samples_B [m];
+			k = (m + dpp->term) & (MAX_TERM - 1);
+		    }
+
+		    left2 = apply_weight (dpp->weight_A, sam_A) + left;
+		    right2 = apply_weight (dpp->weight_B, sam_B) + right;
+
+		    update_weight (dpp->weight_A, dpp->delta, sam_A, left);
+		    update_weight (dpp->weight_B, dpp->delta, sam_B, right);
+
+		    dpp->samples_A [k] = left = left2;
+		    dpp->samples_B [k] = right = right2;
+		}
+		else if (dpp->term == -1) {
+		    left2 = left + apply_weight (dpp->weight_A, dpp->samples_A [0]);
+		    update_weight_clip (dpp->weight_A, dpp->delta, dpp->samples_A [0], left);
+		    left = left2;
+		    right2 = right + apply_weight (dpp->weight_B, left2);
+		    update_weight_clip (dpp->weight_B, dpp->delta, left2, right);
+		    dpp->samples_A [0] = right = right2;
+		}
+		else {
+		    right2 = right + apply_weight (dpp->weight_B, dpp->samples_B [0]);
+		    update_weight_clip (dpp->weight_B, dpp->delta, dpp->samples_B [0], right);
+		    right = right2;
+
+		    if (dpp->term == -3) {
+			right2 = dpp->samples_A [0];
+			dpp->samples_A [0] = right;
+		    }
+
+		    left2 = left + apply_weight (dpp->weight_A, right2);
+		    update_weight_clip (dpp->weight_A, dpp->delta, right2, left);
+		    dpp->samples_B [0] = left = left2;
+		}
+
+	    m = (m + 1) & (MAX_TERM - 1);
+
+	    if (flags & JOINT_STEREO)
+		left += (right -= (left >> 1));
+
+	    if (labs (left) > mute_limit || labs (right) > mute_limit)
+		break;
+
+	    crc = (crc * 3 + left) * 3 + right;
+	    *bptr++ = left;
+	    *bptr++ = right;
+	}
+
+    if (i != sample_count) {
+	memset (buffer, 0, sample_count * (flags & MONO_FLAG ? 4 : 8));
+	wps->mute_error = TRUE;
+	i = sample_count;
+    }
+
+    while (m--)
+	for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
+	    if (dpp->term > 0 && dpp->term <= MAX_TERM) {
+		long temp = dpp->samples_A [0];
+		memcpy (dpp->samples_A, dpp->samples_A + 1, sizeof (dpp->samples_A) - sizeof (dpp->samples_A [0]));
+		dpp->samples_A [MAX_TERM - 1] = temp;
+		temp = dpp->samples_B [0];
+		memcpy (dpp->samples_B, dpp->samples_B + 1, sizeof (dpp->samples_B) - sizeof (dpp->samples_B [0]));
+		dpp->samples_B [MAX_TERM - 1] = temp;
+	    }
+
+    fixup_samples (wps, buffer, i);
+
+    if (flags & FLOAT_DATA)
+	float_normalize (buffer, (flags & MONO_FLAG) ? i : i * 2,
+	    127 - wps->float_norm_exp + wpc->norm_offset);
+
+    wps->sample_index += i;
+    wps->crc = crc;
+
+    return i;
+}
+
+// This is a helper function for unpack_samples() that applies several final
+// operations. First, if the data is 32-bit float data, then that conversion
+// is done in the float.c module (whether lossy or lossless) and we return.
+// Otherwise, if the extended integer data applies, then that operation is
+// executed first. If the unpacked data is lossy (and not corrected) then
+// it is clipped and shifted in a single operation. Otherwise, if it's
+// lossless then the last step is to apply the final shift (if any).
+
+static void fixup_samples (WavpackStream *wps, long *buffer, ulong sample_count)
+{
+    ulong flags = wps->wphdr.flags;
+    int shift = (flags & SHIFT_MASK) >> SHIFT_LSB;
+
+    if (flags & FLOAT_DATA) {
+	float_values (wps, buffer, (flags & MONO_FLAG) ? sample_count : sample_count * 2);
+	return;
+    }
+
+    if (flags & INT32_DATA) {
+	ulong count = (flags & MONO_FLAG) ? sample_count : sample_count * 2;
+	int sent_bits = wps->int32_sent_bits, zeros = wps->int32_zeros;
+	int ones = wps->int32_ones, dups = wps->int32_dups;
+//	ulong mask = (1 << sent_bits) - 1;
+	long *dptr = buffer;
+
+	if (!(flags & HYBRID_FLAG) && !sent_bits && (zeros + ones + dups))
+	    while (count--) {
+		if (zeros)
+		    *dptr <<= zeros;
+		else if (ones)
+		    *dptr = ((*dptr + 1) << ones) - 1;
+		else if (dups)
+		    *dptr = ((*dptr + (*dptr & 1)) << dups) - (*dptr & 1);
+
+		dptr++;
+	    }
+	else
+	    shift += zeros + sent_bits + ones + dups;
+    }
+
+    if (flags & HYBRID_FLAG) {
+	long min_value, max_value, min_shifted, max_shifted;
+
+	switch (flags & BYTES_STORED) {
+	    case 0:
+		min_shifted = (min_value = -128 >> shift) << shift;
+		max_shifted = (max_value = 127 >> shift) << shift;
+		break;
+
+	    case 1:
+		min_shifted = (min_value = -32768 >> shift) << shift;
+		max_shifted = (max_value = 32767 >> shift) << shift;
+		break;
+
+	    case 2:
+		min_shifted = (min_value = -8388608 >> shift) << shift;
+		max_shifted = (max_value = 8388607 >> shift) << shift;
+		break;
+
+	    case 3:
+		min_shifted = (min_value = -(long)2147483648 >> shift) << shift;
+		max_shifted = (max_value = (long) 2147483647 >> shift) << shift;
+		break;
+	}
+
+	if (!(flags & MONO_FLAG))
+	    sample_count *= 2;
+
+	while (sample_count--) {
+	    if (*buffer < min_value)
+		*buffer++ = min_shifted;
+	    else if (*buffer > max_value)
+		*buffer++ = max_shifted;
+	    else
+		*buffer++ <<= shift;
+	}
+    }
+    else if (shift) {
+	if (!(flags & MONO_FLAG))
+	    sample_count *= 2;
+
+	while (sample_count--)
+	    *buffer++ <<= shift;
+    }
+}
+
+// This function checks the crc value(s) for an unpacked block, returning the
+// number of actual crc errors detected for the block. The block must be
+// completely unpacked before this test is valid. For losslessly unpacked
+// blocks of float or extended integer data the extended crc is also checked.
+// Note that WavPack's crc is not a CCITT approved polynomial algorithm, but
+// is a much simpler method that is virtually as robust for real world data.
+
+int check_crc_error (WavpackContext *wpc)
+{
+    WavpackStream *wps = &wpc->stream;
+    int result = 0;
+
+    if (wps->crc != wps->wphdr.crc)
+	++result;
+
+    return result;
+}
diff --git a/apps/codecs/libwavpack/wavpack.h b/apps/codecs/libwavpack/wavpack.h
new file mode 100644
index 0000000..af9d88d
--- /dev/null
+++ b/apps/codecs/libwavpack/wavpack.h
@@ -0,0 +1,330 @@
+////////////////////////////////////////////////////////////////////////////
+//                           **** WAVPACK ****                            //
+//                  Hybrid Lossless Wavefile Compressor                   //
+//              Copyright (c) 1998 - 2004 Conifer Software.               //
+//                          All Rights Reserved.                          //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+// wavpack.h
+
+#include <sys/types.h>
+
+// This header file contains all the definitions required by WavPack.
+
+typedef unsigned char	uchar;
+#if !defined(__GNUC__) || defined(WIN32)
+typedef unsigned short	ushort;
+typedef unsigned long	ulong;
+typedef unsigned int	uint;
+#elif defined(__APPLE__)
+typedef unsigned long	ulong;
+#endif
+
+// This structure is used to access the individual fields of 32-bit ieee
+// floating point numbers. This will not be compatible with compilers that
+// allocate bit fields from the most significant bits, although I'm not sure
+// how common that is.
+
+typedef struct {
+    unsigned mantissa : 23;
+    unsigned exponent : 8;
+    unsigned sign : 1;
+} f32;
+
+#include <stdio.h>
+
+#define FALSE 0
+#define TRUE 1
+
+////////////////////////////// WavPack Header /////////////////////////////////
+
+// Note that this is the ONLY structure that is written to (or read from)
+// WavPack 4.0 files, and is the preamble to every block in both the .wv
+// and .wvc files.
+
+typedef struct {
+    char ckID [4];
+    ulong ckSize;
+    short version;
+    uchar track_no, index_no;
+    ulong total_samples, block_index, block_samples, flags, crc;
+} WavpackHeader;
+
+#define WavpackHeaderFormat "4LS2LLLLL"
+
+// or-values for "flags"
+
+#define BYTES_STORED	3	// 1-4 bytes/sample
+#define MONO_FLAG	4	// not stereo
+#define HYBRID_FLAG	8	// hybrid mode
+#define JOINT_STEREO	0x10	// joint stereo
+#define CROSS_DECORR	0x20	// no-delay cross decorrelation
+#define HYBRID_SHAPE	0x40	// noise shape (hybrid mode only)
+#define FLOAT_DATA	0x80	// ieee 32-bit floating point data
+
+#define INT32_DATA	0x100	// special extended int handling
+#define HYBRID_BITRATE	0x200	// bitrate noise (hybrid mode only)
+#define HYBRID_BALANCE	0x400	// balance noise (hybrid stereo mode only)
+
+#define INITIAL_BLOCK	0x800	// initial block of multichannel segment
+#define FINAL_BLOCK	0x1000	// final block of multichannel segment
+
+#define SHIFT_LSB	13
+#define SHIFT_MASK	(0x1fL << SHIFT_LSB)
+
+#define MAG_LSB		18
+#define MAG_MASK	(0x1fL << MAG_LSB)
+
+#define SRATE_LSB	23
+#define SRATE_MASK	(0xfL << SRATE_LSB)
+
+#define IGNORED_FLAGS	0x18000000	// reserved, but ignore if encountered
+#define NEW_SHAPING	0x20000000	// use IIR filter for negative shaping
+#define UNKNOWN_FLAGS	0xC0000000	// also reserved, but refuse decode if
+					//  encountered
+
+//////////////////////////// WavPack Metadata /////////////////////////////////
+
+// This is an internal representation of metadata.
+
+typedef struct {
+    long byte_length;
+    void *data;
+    uchar id;
+} WavpackMetadata;
+
+#define ID_OPTIONAL_DATA	0x20
+#define ID_ODD_SIZE		0x40
+#define ID_LARGE		0x80
+
+#define ID_DUMMY		0x0
+#define ID_ENCODER_INFO		0x1
+#define ID_DECORR_TERMS		0x2
+#define ID_DECORR_WEIGHTS	0x3
+#define ID_DECORR_SAMPLES	0x4
+#define ID_ENTROPY_VARS		0x5
+#define ID_HYBRID_PROFILE	0x6
+#define ID_SHAPING_WEIGHTS	0x7
+#define ID_FLOAT_INFO		0x8
+#define ID_INT32_INFO		0x9
+#define ID_WV_BITSTREAM		0xa
+#define ID_WVC_BITSTREAM	0xb
+#define ID_WVX_BITSTREAM	0xc
+#define ID_CHANNEL_INFO		0xd
+
+#define ID_RIFF_HEADER		(ID_OPTIONAL_DATA | 0x1)
+#define ID_RIFF_TRAILER		(ID_OPTIONAL_DATA | 0x2)
+#define ID_REPLAY_GAIN		(ID_OPTIONAL_DATA | 0x3)
+#define ID_CUESHEET		(ID_OPTIONAL_DATA | 0x4)
+#define ID_CONFIG_BLOCK		(ID_OPTIONAL_DATA | 0x5)
+#define ID_MD5_CHECKSUM		(ID_OPTIONAL_DATA | 0x6)
+
+///////////////////////// WavPack Configuration ///////////////////////////////
+
+// This internal structure is used during encode to provide configuration to
+// the encoding engine and during decoding to provide fle information back to
+// the higher level functions. Not all fields are used in both modes.
+
+typedef struct {
+    int bits_per_sample, bytes_per_sample;
+    int qmode, flags, xmode, num_channels, float_norm_exp;
+    long block_samples, extra_flags, sample_rate, channel_mask;
+} WavpackConfig;
+
+#define CONFIG_BYTES_STORED	3	// 1-4 bytes/sample
+#define CONFIG_MONO_FLAG	4	// not stereo
+#define CONFIG_HYBRID_FLAG	8	// hybrid mode
+#define CONFIG_JOINT_STEREO	0x10	// joint stereo
+#define CONFIG_CROSS_DECORR	0x20	// no-delay cross decorrelation
+#define CONFIG_HYBRID_SHAPE	0x40	// noise shape (hybrid mode only)
+#define CONFIG_FLOAT_DATA	0x80	// ieee 32-bit floating point data
+
+#define CONFIG_ADOBE_MODE	0x100	// "adobe" mode for 32-bit floats
+#define CONFIG_FAST_FLAG	0x200	// fast mode
+#define CONFIG_VERY_FAST_FLAG	0x400	// double fast
+#define CONFIG_HIGH_FLAG	0x800	// high quality mode
+#define CONFIG_VERY_HIGH_FLAG	0x1000	// double high (not used yet)
+#define CONFIG_BITRATE_KBPS	0x2000	// bitrate is kbps, not bits / sample
+#define CONFIG_AUTO_SHAPING	0x4000	// automatic noise shaping
+#define CONFIG_SHAPE_OVERRIDE	0x8000	// shaping mode specified
+#define CONFIG_JOINT_OVERRIDE	0x10000	// joint-stereo mode specified
+#define CONFIG_COPY_TIME	0x20000	// copy file-time from source
+#define CONFIG_CREATE_EXE	0x40000	// create executable (not yet)
+#define CONFIG_CREATE_WVC	0x80000	// create correction file
+#define CONFIG_OPTIMIZE_WVC	0x100000 // maximize bybrid compression
+#define CONFIG_QUALITY_MODE	0x200000 // psychoacoustic quality mode
+#define CONFIG_RAW_FLAG		0x400000 // raw mode (not implemented yet)
+#define CONFIG_CALC_NOISE	0x800000 // calc noise in hybrid mode
+#define CONFIG_LOSSY_MODE	0x1000000 // obsolete (for information)
+#define CONFIG_EXTRA_MODE	0x2000000 // extra processing mode
+#define CONFIG_SKIP_WVX		0x4000000 // no wvx stream w/ floats & big ints
+#define CONFIG_MD5_CHECKSUM	0x8000000 // compute & store MD5 signature
+#define CONFIG_QUIET_MODE	0x10000000 // don't report progress %
+
+//////////////////////////////// WavPack Stream ///////////////////////////////
+
+// This internal structure contains everything required to handle a WavPack
+// "stream", which is defined as a stereo or mono stream of audio samples. For
+// multichannel audio several of these would be required. Each stream contains
+// pointers to hold a complete allocated block of WavPack data, although it's
+// possible to decode WavPack blocks without buffering an entire block.
+
+typedef long (*read_stream)(void *, long);
+
+typedef struct bs {
+    uchar *buf, *end, *ptr;
+    void (*wrap)(struct bs *bs);
+    ulong file_bytes, sr;
+    int error, bc;
+    read_stream file;
+} Bitstream;
+
+#define MAX_NTERMS 16
+#define MAX_TERM 8
+
+struct decorr_pass {
+    short term, delta, weight_A, weight_B;
+    long samples_A [MAX_TERM], samples_B [MAX_TERM];
+};
+
+typedef struct {
+    WavpackHeader wphdr;
+
+    int num_terms, mute_error;
+    ulong sample_index, crc;
+    Bitstream wvbits;
+
+    uchar int32_sent_bits, int32_zeros, int32_ones, int32_dups;
+    uchar float_flags, float_shift, float_max_exp, float_norm_exp;
+ 
+    struct decorr_pass decorr_passes [MAX_NTERMS];
+
+    struct {
+	ulong bitrate_delta [2], bitrate_acc [2];
+	ulong median [3] [2], slow_level [2], error_limit [2];
+	ulong pend_data, holding_one, zeros_acc;
+	int holding_zero, pend_count;
+    } w;
+} WavpackStream;
+
+// flags for float_flags:
+
+#define FLOAT_SHIFT_ONES 1	// bits left-shifted into float = '1'
+#define FLOAT_SHIFT_SAME 2	// bits left-shifted into float are the same
+#define FLOAT_SHIFT_SENT 4	// bits shifted into float are sent literally
+#define FLOAT_ZEROS_SENT 8	// "zeros" are not all real zeros
+#define FLOAT_NEG_ZEROS  0x10	// contains negative zeros
+#define FLOAT_EXCEPTIONS 0x20	// contains exceptions (inf, nan, etc.)
+
+/////////////////////////////// WavPack Context ///////////////////////////////
+
+// This internal structure holds everything required to encode or decode WavPack
+// files. It is recommended that direct access to this structure be minimized
+// and the provided utilities used instead.
+
+typedef struct {
+    WavpackConfig config;
+    WavpackStream stream;
+
+    uchar read_buffer [1024];
+    char error_message [80];
+
+    read_stream infile;
+    ulong total_samples, crc_errors, first_flags;
+    int open_flags, norm_offset, reduced_channels, lossy_blocks;
+
+} WavpackContext;
+
+//////////////////////// function prototypes and macros //////////////////////
+
+#define CLEAR(destin) memset (&destin, 0, sizeof (destin));
+
+// bits.c
+
+void bs_open_read (Bitstream *bs, uchar *buffer_start, uchar *buffer_end, read_stream file, ulong file_bytes);
+
+#define bs_is_open(bs) ((bs)->ptr != NULL)
+
+#define getbit(bs) ( \
+    (((bs)->bc) ? \
+	((bs)->bc--, (bs)->sr & 1) : \
+	    (((++((bs)->ptr) != (bs)->end) ? (void) 0 : (bs)->wrap (bs)), (bs)->bc = 7, ((bs)->sr = *((bs)->ptr)) & 1) \
+    ) ? \
+	((bs)->sr >>= 1, 1) : \
+	((bs)->sr >>= 1, 0) \
+)
+
+#define getbits(value, nbits, bs) { \
+    while ((nbits) > (bs)->bc) { \
+	if (++((bs)->ptr) == (bs)->end) (bs)->wrap (bs); \
+	(bs)->sr |= (long)*((bs)->ptr) << (bs)->bc; \
+	(bs)->bc += 8; \
+    } \
+    *(value) = (bs)->sr; \
+    (bs)->sr >>= (nbits); \
+    (bs)->bc -= (nbits); \
+}
+
+void little_endian_to_native (void *data, char *format);
+void native_to_little_endian (void *data, char *format);
+
+// unpack.c
+
+int unpack_init (WavpackContext *wpc);
+int init_wv_bitstream (WavpackContext *wpc, WavpackMetadata *wpmd);
+int read_decorr_terms (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_decorr_weights (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_decorr_samples (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_float_info (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_int32_info (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_channel_info (WavpackContext *wpc, WavpackMetadata *wpmd);
+int read_config_info (WavpackContext *wpc, WavpackMetadata *wpmd);
+long unpack_samples (WavpackContext *wpc, long *buffer, ulong sample_count);
+int check_crc_error (WavpackContext *wpc);
+
+// metadata.c stuff
+
+int read_metadata_buff (WavpackContext *wpc, WavpackMetadata *wpmd);
+int process_metadata (WavpackContext *wpc, WavpackMetadata *wpmd);
+
+// words.c stuff
+
+int read_entropy_vars (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_hybrid_profile (WavpackStream *wps, WavpackMetadata *wpmd);
+long get_word (WavpackStream *wps, int chan);
+long exp2s (int log);
+int restore_weight (char weight);
+
+#define WORD_EOF (1L << 31)
+
+// float.c
+
+int read_float_info (WavpackStream *wps, WavpackMetadata *wpmd);
+void float_values (WavpackStream *wps, long *values, long num_values);
+void float_normalize (long *values, long num_values, int delta_exp);
+
+// wputils.c
+
+WavpackContext *WavpackOpenFileInput (read_stream infile, char *error);
+
+int WavpackGetMode (WavpackContext *wpc);
+
+#define MODE_WVC	0x1
+#define MODE_LOSSLESS	0x2
+#define MODE_HYBRID	0x4
+#define MODE_FLOAT	0x8
+#define MODE_VALID_TAG	0x10
+#define MODE_HIGH	0x20
+#define MODE_FAST	0x40
+
+ulong WavpackUnpackSamples (WavpackContext *wpc, long *buffer, ulong samples);
+ulong WavpackGetNumSamples (WavpackContext *wpc);
+ulong WavpackGetSampleIndex (WavpackContext *wpc);
+int WavpackGetNumErrors (WavpackContext *wpc);
+int WavpackLossyBlocks (WavpackContext *wpc);
+ulong WavpackGetSampleRate (WavpackContext *wpc);
+int WavpackGetBitsPerSample (WavpackContext *wpc);
+int WavpackGetBytesPerSample (WavpackContext *wpc);
+int WavpackGetNumChannels (WavpackContext *wpc);
+int WavpackGetReducedChannels (WavpackContext *wpc);
diff --git a/apps/codecs/libwavpack/words.c b/apps/codecs/libwavpack/words.c
new file mode 100644
index 0000000..35061b6
--- /dev/null
+++ b/apps/codecs/libwavpack/words.c
@@ -0,0 +1,503 @@
+////////////////////////////////////////////////////////////////////////////
+//                           **** WAVPACK ****                            //
+//                  Hybrid Lossless Wavefile Compressor                   //
+//              Copyright (c) 1998 - 2004 Conifer Software.               //
+//                          All Rights Reserved.                          //
+////////////////////////////////////////////////////////////////////////////
+
+// words.c
+
+// This module provides entropy word encoding and decoding functions using
+// a variation on the Rice method.  This was introduced in version 3.93
+// because it allows splitting the data into a "lossy" stream and a
+// "correction" stream in a very efficient manner and is therefore ideal
+// for the "hybrid" mode.  For 4.0, the efficiency of this method was
+// significantly improved by moving away from the normal Rice restriction of
+// using powers of two for the modulus divisions and now the method can be
+// used for both hybrid and pure lossless encoding.
+
+// Samples are divided by median probabilities at 5/7 (71.43%), 10/49 (20.41%),
+// and 20/343 (5.83%). Each zone has 3.5 times fewer samples than the
+// previous. Using standard Rice coding on this data would result in 1.4
+// bits per sample average (not counting sign bit). However, there is a
+// very simple encoding that is over 99% efficient with this data and
+// results in about 1.22 bits per sample.
+
+#include "wavpack.h"
+
+#include <string.h>
+
+//////////////////////////////// local macros /////////////////////////////////
+
+#define LIMIT_ONES 16	// maximum consecutive 1s sent for "div" data
+
+// these control the time constant "slow_level" which is used for hybrid mode
+// that controls bitrate as a function of residual level (HYBRID_BITRATE).
+#define SLS 8
+#define SLO ((1 << (SLS - 1)))
+
+// these control the time constant of the 3 median level breakpoints
+#define DIV0 128	// 5/7 of samples
+#define DIV1 64		// 10/49 of samples
+#define DIV2 32		// 20/343 of samples
+
+// this macro retrieves the specified median breakpoint (without frac; min = 1)
+#define GET_MED(med) (((wps->w.median [med] [chan]) >> 4) + 1)
+
+// These macros update the specified median breakpoints. Note that the median
+// is incremented when the sample is higher than the median, else decremented.
+// They are designed so that the median will never drop below 1 and the value
+// is essentially stationary if there are 2 increments for every 5 decrements.
+
+#define INC_MED0() (wps->w.median [0] [chan] += ((wps->w.median [0] [chan] + DIV0) / DIV0) * 5)
+#define DEC_MED0() (wps->w.median [0] [chan] -= ((wps->w.median [0] [chan] + (DIV0-2)) / DIV0) * 2)
+#define INC_MED1() (wps->w.median [1] [chan] += ((wps->w.median [1] [chan] + DIV1) / DIV1) * 5)
+#define DEC_MED1() (wps->w.median [1] [chan] -= ((wps->w.median [1] [chan] + (DIV1-2)) / DIV1) * 2)
+#define INC_MED2() (wps->w.median [2] [chan] += ((wps->w.median [2] [chan] + DIV2) / DIV2) * 5)
+#define DEC_MED2() (wps->w.median [2] [chan] -= ((wps->w.median [2] [chan] + (DIV2-2)) / DIV2) * 2)
+
+#define count_bits(av) ( \
+ (av) < (1 << 8) ? nbits_table [av] : \
+  ( \
+   (av) < (1L << 16) ? nbits_table [(av) >> 8] + 8 : \
+   ((av) < (1L << 24) ? nbits_table [(av) >> 16] + 16 : nbits_table [(av) >> 24] + 24) \
+  ) \
+)
+
+///////////////////////////// local table storage ////////////////////////////
+
+const char nbits_table [] = {
+    0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4,	// 0 - 15
+    5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,	// 16 - 31
+    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,	// 32 - 47
+    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,	// 48 - 63
+    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,	// 64 - 79
+    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,	// 80 - 95
+    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,	// 96 - 111
+    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,	// 112 - 127
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,	// 128 - 143
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,	// 144 - 159
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,	// 160 - 175
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,	// 176 - 191
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,	// 192 - 207
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,	// 208 - 223
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,	// 224 - 239
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8	// 240 - 255
+};
+
+static const uchar log2_table [] = {
+    0x00, 0x01, 0x03, 0x04, 0x06, 0x07, 0x09, 0x0a, 0x0b, 0x0d, 0x0e, 0x10, 0x11, 0x12, 0x14, 0x15,
+    0x16, 0x18, 0x19, 0x1a, 0x1c, 0x1d, 0x1e, 0x20, 0x21, 0x22, 0x24, 0x25, 0x26, 0x28, 0x29, 0x2a,
+    0x2c, 0x2d, 0x2e, 0x2f, 0x31, 0x32, 0x33, 0x34, 0x36, 0x37, 0x38, 0x39, 0x3b, 0x3c, 0x3d, 0x3e,
+    0x3f, 0x41, 0x42, 0x43, 0x44, 0x45, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4d, 0x4e, 0x4f, 0x50, 0x51,
+    0x52, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5a, 0x5c, 0x5d, 0x5e, 0x5f, 0x60, 0x61, 0x62, 0x63,
+    0x64, 0x66, 0x67, 0x68, 0x69, 0x6a, 0x6b, 0x6c, 0x6d, 0x6e, 0x6f, 0x70, 0x71, 0x72, 0x74, 0x75,
+    0x76, 0x77, 0x78, 0x79, 0x7a, 0x7b, 0x7c, 0x7d, 0x7e, 0x7f, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85,
+    0x86, 0x87, 0x88, 0x89, 0x8a, 0x8b, 0x8c, 0x8d, 0x8e, 0x8f, 0x90, 0x91, 0x92, 0x93, 0x94, 0x95,
+    0x96, 0x97, 0x98, 0x99, 0x9a, 0x9b, 0x9b, 0x9c, 0x9d, 0x9e, 0x9f, 0xa0, 0xa1, 0xa2, 0xa3, 0xa4,
+    0xa5, 0xa6, 0xa7, 0xa8, 0xa9, 0xa9, 0xaa, 0xab, 0xac, 0xad, 0xae, 0xaf, 0xb0, 0xb1, 0xb2, 0xb2,
+    0xb3, 0xb4, 0xb5, 0xb6, 0xb7, 0xb8, 0xb9, 0xb9, 0xba, 0xbb, 0xbc, 0xbd, 0xbe, 0xbf, 0xc0, 0xc0,
+    0xc1, 0xc2, 0xc3, 0xc4, 0xc5, 0xc6, 0xc6, 0xc7, 0xc8, 0xc9, 0xca, 0xcb, 0xcb, 0xcc, 0xcd, 0xce,
+    0xcf, 0xd0, 0xd0, 0xd1, 0xd2, 0xd3, 0xd4, 0xd4, 0xd5, 0xd6, 0xd7, 0xd8, 0xd8, 0xd9, 0xda, 0xdb,
+    0xdc, 0xdc, 0xdd, 0xde, 0xdf, 0xe0, 0xe0, 0xe1, 0xe2, 0xe3, 0xe4, 0xe4, 0xe5, 0xe6, 0xe7, 0xe7,
+    0xe8, 0xe9, 0xea, 0xea, 0xeb, 0xec, 0xed, 0xee, 0xee, 0xef, 0xf0, 0xf1, 0xf1, 0xf2, 0xf3, 0xf4,
+    0xf4, 0xf5, 0xf6, 0xf7, 0xf7, 0xf8, 0xf9, 0xf9, 0xfa, 0xfb, 0xfc, 0xfc, 0xfd, 0xfe, 0xff, 0xff
+};
+
+static const uchar exp2_table [] = {
+    0x00, 0x01, 0x01, 0x02, 0x03, 0x03, 0x04, 0x05, 0x06, 0x06, 0x07, 0x08, 0x08, 0x09, 0x0a, 0x0b,
+    0x0b, 0x0c, 0x0d, 0x0e, 0x0e, 0x0f, 0x10, 0x10, 0x11, 0x12, 0x13, 0x13, 0x14, 0x15, 0x16, 0x16,
+    0x17, 0x18, 0x19, 0x19, 0x1a, 0x1b, 0x1c, 0x1d, 0x1d, 0x1e, 0x1f, 0x20, 0x20, 0x21, 0x22, 0x23,
+    0x24, 0x24, 0x25, 0x26, 0x27, 0x28, 0x28, 0x29, 0x2a, 0x2b, 0x2c, 0x2c, 0x2d, 0x2e, 0x2f, 0x30,
+    0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3a, 0x3a, 0x3b, 0x3c, 0x3d,
+    0x3e, 0x3f, 0x40, 0x41, 0x41, 0x42, 0x43, 0x44, 0x45, 0x46, 0x47, 0x48, 0x48, 0x49, 0x4a, 0x4b,
+    0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5a,
+    0x5b, 0x5c, 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66, 0x67, 0x68, 0x69,
+    0x6a, 0x6b, 0x6c, 0x6d, 0x6e, 0x6f, 0x70, 0x71, 0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79,
+    0x7a, 0x7b, 0x7c, 0x7d, 0x7e, 0x7f, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x87, 0x88, 0x89, 0x8a,
+    0x8b, 0x8c, 0x8d, 0x8e, 0x8f, 0x90, 0x91, 0x92, 0x93, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9a, 0x9b,
+    0x9c, 0x9d, 0x9f, 0xa0, 0xa1, 0xa2, 0xa3, 0xa4, 0xa5, 0xa6, 0xa8, 0xa9, 0xaa, 0xab, 0xac, 0xad,
+    0xaf, 0xb0, 0xb1, 0xb2, 0xb3, 0xb4, 0xb6, 0xb7, 0xb8, 0xb9, 0xba, 0xbc, 0xbd, 0xbe, 0xbf, 0xc0,
+    0xc2, 0xc3, 0xc4, 0xc5, 0xc6, 0xc8, 0xc9, 0xca, 0xcb, 0xcd, 0xce, 0xcf, 0xd0, 0xd2, 0xd3, 0xd4,
+    0xd6, 0xd7, 0xd8, 0xd9, 0xdb, 0xdc, 0xdd, 0xde, 0xe0, 0xe1, 0xe2, 0xe4, 0xe5, 0xe6, 0xe8, 0xe9,
+    0xea, 0xec, 0xed, 0xee, 0xf0, 0xf1, 0xf2, 0xf4, 0xf5, 0xf6, 0xf8, 0xf9, 0xfa, 0xfc, 0xfd, 0xff
+};
+
+///////////////////////////// executable code ////////////////////////////////
+
+static int log2 (unsigned long avalue);
+
+// Read the median log2 values from the specifed metadata structure, convert
+// them back to 32-bit unsigned values and store them. If length is not
+// exactly correct then we flag and return an error.
+
+int read_entropy_vars (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+    uchar *byteptr = wpmd->data;
+
+    if (wpmd->byte_length != ((wps->wphdr.flags & MONO_FLAG) ? 6 : 12))
+	return FALSE;
+
+    wps->w.median [0] [0] = exp2s (byteptr [0] + (byteptr [1] << 8));
+    wps->w.median [1] [0] = exp2s (byteptr [2] + (byteptr [3] << 8));
+    wps->w.median [2] [0] = exp2s (byteptr [4] + (byteptr [5] << 8));
+
+    if (!(wps->wphdr.flags & MONO_FLAG)) {
+	wps->w.median [0] [1] = exp2s (byteptr [6] + (byteptr [7] << 8));
+	wps->w.median [1] [1] = exp2s (byteptr [8] + (byteptr [9] << 8));
+	wps->w.median [2] [1] = exp2s (byteptr [10] + (byteptr [11] << 8));
+    }
+
+    return TRUE;
+}
+
+// Read the hybrid related values from the specifed metadata structure, convert
+// them back to their internal formats and store them. The extended profile
+// stuff is not implemented yet, so return an error if we get more data than
+// we know what to do with.
+
+int read_hybrid_profile (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+    uchar *byteptr = wpmd->data;
+    uchar *endptr = byteptr + wpmd->byte_length;
+
+    if (wps->wphdr.flags & HYBRID_BITRATE) {
+	wps->w.slow_level [0] = exp2s (byteptr [0] + (byteptr [1] << 8));
+	byteptr += 2;
+
+	if (!(wps->wphdr.flags & MONO_FLAG)) {
+	    wps->w.slow_level [1] = exp2s (byteptr [0] + (byteptr [1] << 8));
+	    byteptr += 2;
+	}
+    }
+
+    wps->w.bitrate_acc [0] = (long)(byteptr [0] + (byteptr [1] << 8)) << 16;
+    byteptr += 2;
+
+    if (!(wps->wphdr.flags & MONO_FLAG)) {
+	wps->w.bitrate_acc [1] = (long)(byteptr [0] + (byteptr [1] << 8)) << 16;
+	byteptr += 2;
+    }
+
+    if (byteptr < endptr) {
+	wps->w.bitrate_delta [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+	byteptr += 2;
+
+	if (!(wps->wphdr.flags & MONO_FLAG)) {
+	    wps->w.bitrate_delta [1] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+	    byteptr += 2;
+	}
+
+	if (byteptr < endptr)
+	    return FALSE;
+    }
+    else
+	wps->w.bitrate_delta [0] = wps->w.bitrate_delta [1] = 0;
+
+    return TRUE;
+}
+
+// This function is called during both encoding and decoding of hybrid data to
+// update the "error_limit" variable which determines the maximum sample error
+// allowed in the main bitstream. In the HYBRID_BITRATE mode (which is the only
+// currently implemented) this is calculated from the slow_level values and the
+// bitrate accumulators. Note that the bitrate accumulators can be changing.
+
+static void update_error_limit (WavpackStream *wps)
+{
+    int bitrate_0 = (wps->w.bitrate_acc [0] += wps->w.bitrate_delta [0]) >> 16;
+
+    if (wps->wphdr.flags & MONO_FLAG) {
+	if (wps->wphdr.flags & HYBRID_BITRATE) {
+	    int slow_log_0 = (wps->w.slow_level [0] + SLO) >> SLS;
+
+	    if (slow_log_0 - bitrate_0 > -0x100)
+		wps->w.error_limit [0] = exp2s (slow_log_0 - bitrate_0 + 0x100);
+	    else
+		wps->w.error_limit [0] = 0;
+	}
+	else
+	    wps->w.error_limit [0] = exp2s (bitrate_0);
+    }
+    else {
+	int bitrate_1 = (wps->w.bitrate_acc [1] += wps->w.bitrate_delta [1]) >> 16;
+
+	if (wps->wphdr.flags & HYBRID_BITRATE) {
+	    int slow_log_0 = (wps->w.slow_level [0] + SLO) >> SLS;
+	    int slow_log_1 = (wps->w.slow_level [1] + SLO) >> SLS;
+
+	    if (wps->wphdr.flags & HYBRID_BALANCE) {
+		int balance = (slow_log_1 - slow_log_0 + bitrate_1 + 1) >> 1;
+
+		if (balance > bitrate_0) {
+		    bitrate_1 = bitrate_0 * 2;
+		    bitrate_0 = 0;
+		}
+		else if (-balance > bitrate_0) {
+		    bitrate_0 = bitrate_0 * 2;
+		    bitrate_1 = 0;
+		}
+		else {
+		    bitrate_1 = bitrate_0 + balance;
+		    bitrate_0 = bitrate_0 - balance;
+		}
+	    }
+
+	    if (slow_log_0 - bitrate_0 > -0x100)
+		wps->w.error_limit [0] = exp2s (slow_log_0 - bitrate_0 + 0x100);
+	    else
+		wps->w.error_limit [0] = 0;
+
+	    if (slow_log_1 - bitrate_1 > -0x100)
+		wps->w.error_limit [1] = exp2s (slow_log_1 - bitrate_1 + 0x100);
+	    else
+		wps->w.error_limit [1] = 0;
+	}
+	else {
+	    wps->w.error_limit [0] = exp2s (bitrate_0);
+	    wps->w.error_limit [1] = exp2s (bitrate_1);
+	}
+    }
+}
+
+static ulong read_code (Bitstream *bs, ulong maxcode);
+
+// Read the next word from the bitstream "wvbits" and return the value. This
+// function can be used for hybrid or lossless streams, but since an
+// optimized version is available for lossless this function would normally
+// be used for hybrid only. If a hybrid lossless stream is being read then
+// the "correction" offset is written at the specified pointer. A return value
+// of WORD_EOF indicates that the end of the bitstream was reached (all 1s) or
+// some other error occurred.
+
+long get_word (WavpackStream *wps, int chan)
+{
+    ulong ones_count, low, mid, high;
+    int sign;
+
+    if (wps->w.zeros_acc) {
+	if (--wps->w.zeros_acc) {
+	    wps->w.slow_level [chan] -= (wps->w.slow_level [chan] + SLO) >> SLS;
+	    return 0;
+	}
+    }
+    else if (!wps->w.holding_zero && !wps->w.holding_one && !(wps->w.median [0] [0] & ~1) && !(wps->w.median [0] [1] & ~1)) {
+	ulong mask;
+	int cbits;
+
+	for (cbits = 0; cbits < 33 && getbit (&wps->wvbits); ++cbits);
+
+	if (cbits == 33)
+	    return WORD_EOF;
+
+	if (cbits < 2)
+	    wps->w.zeros_acc = cbits;
+	else {
+	    for (mask = 1, wps->w.zeros_acc = 0; --cbits; mask <<= 1)
+		if (getbit (&wps->wvbits))
+		    wps->w.zeros_acc |= mask;
+
+	    wps->w.zeros_acc |= mask;
+	}
+
+	if (wps->w.zeros_acc) {
+	    wps->w.slow_level [chan] -= (wps->w.slow_level [chan] + SLO) >> SLS;
+	    CLEAR (wps->w.median);
+	    return 0;
+	}
+    }
+
+    if (wps->w.holding_zero)
+	ones_count = wps->w.holding_zero = 0;
+    else {
+#ifdef LIMIT_ONES
+	for (ones_count = 0; ones_count < (LIMIT_ONES + 1) && getbit (&wps->wvbits); ++ones_count);
+
+	if (ones_count == (LIMIT_ONES + 1))
+	    return WORD_EOF;
+
+	if (ones_count == LIMIT_ONES) {
+	    ulong mask;
+	    int cbits;
+
+	    for (cbits = 0; cbits < 33 && getbit (&wps->wvbits); ++cbits);
+
+	    if (cbits == 33)
+		return WORD_EOF;
+
+	    if (cbits < 2)
+		ones_count = cbits;
+	    else {
+		for (mask = 1, ones_count = 0; --cbits; mask <<= 1)
+		    if (getbit (&wps->wvbits))
+			ones_count |= mask;
+
+		ones_count |= mask;
+	    }
+
+	    ones_count += LIMIT_ONES;
+	}
+#else
+	for (ones_count = 0; getbit (&wps->wvbits); ++ones_count);
+#endif
+
+	if (wps->w.holding_one) {
+	    wps->w.holding_one = ones_count & 1;
+	    ones_count = (ones_count >> 1) + 1;
+	}
+	else {
+	    wps->w.holding_one = ones_count & 1;
+	    ones_count >>= 1;
+	}
+
+	wps->w.holding_zero = ~wps->w.holding_one & 1;
+    }
+
+    if ((wps->wphdr.flags & HYBRID_FLAG) && !chan)
+	update_error_limit (wps);
+
+    if (ones_count == 0) {
+	low = 0;
+	high = GET_MED (0) - 1;
+	DEC_MED0 ();
+    }
+    else {
+	low = GET_MED (0);
+	INC_MED0 ();
+
+	if (ones_count == 1) {
+	    high = low + GET_MED (1) - 1;
+	    DEC_MED1 ();
+	}
+	else {
+	    low += GET_MED (1);
+	    INC_MED1 ();
+
+	    if (ones_count == 2) {
+		high = low + GET_MED (2) - 1;
+		DEC_MED2 ();
+	    }
+	    else {
+		low += (ones_count - 2) * GET_MED (2);
+		high = low + GET_MED (2) - 1;
+		INC_MED2 ();
+	    }
+	}
+    }
+
+    mid = (high + low + 1) >> 1;
+
+    if (!wps->w.error_limit [chan])
+	mid = read_code (&wps->wvbits, high - low) + low;
+    else while (high - low > wps->w.error_limit [chan]) {
+	if (getbit (&wps->wvbits))
+	    mid = (high + (low = mid) + 1) >> 1;
+	else
+	    mid = ((high = mid - 1) + low + 1) >> 1;
+    }
+
+    sign = getbit (&wps->wvbits);
+
+    if (wps->wphdr.flags & HYBRID_BITRATE) {
+	wps->w.slow_level [chan] -= (wps->w.slow_level [chan] + SLO) >> SLS;
+	wps->w.slow_level [chan] += log2 (mid);
+    }
+
+    return sign ? ~mid : mid;
+}
+
+// Read a single unsigned value from the specified bitstream with a value
+// from 0 to maxcode. If there are exactly a power of two number of possible
+// codes then this will read a fixed number of bits; otherwise it reads the
+// minimum number of bits and then determines whether another bit is needed
+// to define the code.
+
+static ulong read_code (Bitstream *bs, ulong maxcode)
+{
+    int bitcount = count_bits (maxcode);
+    ulong extras = (1L << bitcount) - maxcode - 1, code;
+
+    if (!bitcount)
+	return 0;
+
+    getbits (&code, bitcount - 1, bs);
+    code &= (1L << (bitcount - 1)) - 1;
+
+    if (code >= extras) {
+	code = (code << 1) - extras;
+
+	if (getbit (bs))
+	    ++code;
+    }
+
+    return code;
+}
+
+// The concept of a base 2 logarithm is used in many parts of WavPack. It is
+// a way of sufficiently accurately representing 32-bit signed and unsigned
+// values storing only 16 bits (actually fewer). It is also used in the hybrid
+// mode for quickly comparing the relative magnitude of large values (i.e.
+// division) and providing smooth exponentials using only addition.
+
+// These are not strict logarithms in that they become linear around zero and
+// can therefore represent both zero and negative values. They have 8 bits
+// of precision and in "roundtrip" conversions the total error never exceeds 1
+// part in 225 except for the cases of +/-115 and +/-195 (which error by 1).
+
+
+// This function returns the log2 for the specified 32-bit unsigned value.
+// The maximum value allowed is about 0xff800000 and returns 8447.
+
+static int log2 (unsigned long avalue)
+{
+    int dbits;
+
+    if ((avalue += avalue >> 9) < (1 << 8)) {
+	dbits = nbits_table [avalue];
+	return (dbits << 8) + log2_table [(avalue << (9 - dbits)) & 0xff];
+    }
+    else {
+	if (avalue < (1L << 16))
+	    dbits = nbits_table [avalue >> 8] + 8;
+	else if (avalue < (1L << 24))
+	    dbits = nbits_table [avalue >> 16] + 16;
+	else
+	    dbits = nbits_table [avalue >> 24] + 24;
+
+	return (dbits << 8) + log2_table [(avalue >> (dbits - 9)) & 0xff];
+    }
+}
+
+// This function returns the original integer represented by the supplied
+// logarithm (at least within the provided accuracy). The log is signed,
+// but since a full 32-bit value is returned this can be used for unsigned
+// conversions as well (i.e. the input range is -8192 to +8447).
+
+long exp2s (int log)
+{
+    ulong value;
+
+    if (log < 0)
+	return -exp2s (-log);
+
+    value = exp2_table [log & 0xff] | 0x100;
+
+    if ((log >>= 8) <= 9)
+	return value >> (9 - log);
+    else
+	return value << (log - 9);
+}
+
+// These two functions convert internal weights (which are normally +/-1024)
+// to and from an 8-bit signed character version for storage in metadata. The
+// weights are clipped here in the case that they are outside that range.
+
+int restore_weight (char weight)
+{
+    int result;
+
+    if ((result = (int) weight << 3) > 0)
+	result += (result + 64) >> 7;
+
+    return result;
+}
diff --git a/apps/codecs/libwavpack/wputils.c b/apps/codecs/libwavpack/wputils.c
new file mode 100644
index 0000000..21c399f
--- /dev/null
+++ b/apps/codecs/libwavpack/wputils.c
@@ -0,0 +1,354 @@
+////////////////////////////////////////////////////////////////////////////
+//			     **** WAVPACK ****				  //
+//		    Hybrid Lossless Wavefile Compressor			  //
+//		Copyright (c) 1998 - 2004 Conifer Software.		  //
+//			    All Rights Reserved.			  //
+//      Distributed under the BSD Software License (see license.txt)      //
+////////////////////////////////////////////////////////////////////////////
+
+// wputils.c
+
+// This module provides a high-level interface for decoding WavPack 4.0 audio
+// streams and files. WavPack data is read with a stream reading callback. No
+// direct seeking is provided for, but it is possible to start decoding
+// anywhere in a WavPack stream. In this case, WavPack will be able to provide
+// the sample-accurate position when it synchs with the data and begins
+// decoding.
+
+#include "wavpack.h"
+
+#include <string.h>
+
+///////////////////////////// local table storage ////////////////////////////
+
+const ulong sample_rates [] = { 6000, 8000, 9600, 11025, 12000, 16000, 22050,
+    24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000 };
+
+///////////////////////////// executable code ////////////////////////////////
+
+static ulong read_next_header (read_stream infile, WavpackHeader *wphdr);
+	
+// This function reads data from the specified stream in search of a valid
+// WavPack 4.0 audio block. If this fails in 1 megabyte (or an invalid or
+// unsupported WavPack block is encountered) then an appropriate message is
+// copied to "error" and NULL is returned, otherwise a pointer to a
+// WavpackContext structure is returned (which is used to call all other
+// functions in this module). This can be initiated at the beginning of a
+// WavPack file, or anywhere inside a WavPack file. To determine the exact
+// position within the file use WavpackGetSampleIndex(). For demonstration
+// purposes this uses a single static copy of the WavpackContext structure,
+// so obviously it cannot be used for more than one file at a time. Also,
+// this function will not handle "correction" files, plays only the first
+// two channels of multi-channel files, and is limited in resolution in some
+// large integer or floating point files (but always provides at least 24 bits
+// of resolution).
+
+static WavpackContext wpc;
+
+WavpackContext *WavpackOpenFileInput (read_stream infile, char *error)
+{
+    WavpackStream *wps = &wpc.stream;
+    ulong bcount;
+
+    CLEAR (wpc);
+    wpc.infile = infile;
+    wpc.total_samples = (ulong) -1;
+    wpc.norm_offset = 0;
+    wpc.open_flags = 0;
+
+    // open the source file for reading and store the size
+
+    while (!wps->wphdr.block_samples) {
+
+	bcount = read_next_header (wpc.infile, &wps->wphdr);
+
+	if (bcount == (ulong) -1) {
+	    strcpy (error, "not compatible with this version of WavPack file!");
+	    return NULL;
+	}
+
+	if ((wps->wphdr.flags & UNKNOWN_FLAGS) || wps->wphdr.version < 0x402 || wps->wphdr.version > 0x40f) {
+	    strcpy (error, "not compatible with this version of WavPack file!");
+	    return NULL;
+	}
+
+	if (wps->wphdr.block_samples && wps->wphdr.total_samples != (ulong) -1)
+	    wpc.total_samples = wps->wphdr.total_samples;
+
+	if (!unpack_init (&wpc)) {
+	    strcpy (error, wpc.error_message [0] ? wpc.error_message :
+		"not compatible with this version of WavPack file!");
+
+	    return NULL;
+	}
+    }
+
+    wpc.config.flags &= ~0xff;
+    wpc.config.flags |= wps->wphdr.flags & 0xff;
+    wpc.config.bytes_per_sample = (wps->wphdr.flags & BYTES_STORED) + 1;
+    wpc.config.float_norm_exp = wps->float_norm_exp;
+
+    wpc.config.bits_per_sample = (wpc.config.bytes_per_sample * 8) - 
+	((wps->wphdr.flags & SHIFT_MASK) >> SHIFT_LSB);
+
+    if (!wpc.config.sample_rate) {
+	if (!wps || !wps->wphdr.block_samples || (wps->wphdr.flags & SRATE_MASK) == SRATE_MASK)
+	    wpc.config.sample_rate = 44100;
+	else
+	    wpc.config.sample_rate = sample_rates [(wps->wphdr.flags & SRATE_MASK) >> SRATE_LSB];
+    }
+
+    if (!wpc.config.num_channels) {
+	wpc.config.num_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
+	wpc.config.channel_mask = 0x5 - wpc.config.num_channels;
+    }
+
+    if (!(wps->wphdr.flags & FINAL_BLOCK))
+	wpc.reduced_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
+
+    return &wpc;
+}
+
+// This function obtains general information about an open file and returns
+// a mask with the following bit values:
+
+// MODE_LOSSLESS:  file is lossless (pure lossless only)
+// MODE_HYBRID:  file is hybrid mode (lossy part only)
+// MODE_FLOAT:  audio data is 32-bit ieee floating point
+// MODE_HIGH:  file was created in "high" mode (information only)
+// MODE_FAST:  file was created in "fast" mode (information only)
+
+int WavpackGetMode (WavpackContext *wpc)
+{
+    int mode = 0;
+
+    if (wpc) {
+	if (wpc->config.flags & CONFIG_HYBRID_FLAG)
+	    mode |= MODE_HYBRID;
+	else if (!(wpc->config.flags & CONFIG_LOSSY_MODE))
+	    mode |= MODE_LOSSLESS;
+
+	if (wpc->lossy_blocks)
+	    mode &= ~MODE_LOSSLESS;
+
+	if (wpc->config.flags & CONFIG_FLOAT_DATA)
+	    mode |= MODE_FLOAT;
+
+	if (wpc->config.flags & CONFIG_HIGH_FLAG)
+	    mode |= MODE_HIGH;
+
+	if (wpc->config.flags & CONFIG_FAST_FLAG)
+	    mode |= MODE_FAST;
+    }
+
+    return mode;
+}
+
+// Unpack the specified number of samples from the current file position.
+// Note that "samples" here refers to "complete" samples, which would be
+// 2 longs for stereo files. The audio data is returned right-justified in
+// 32-bit longs in the endian mode native to the executing processor. So,
+// if the original data was 16-bit, then the values returned would be
+// +/-32k. Floating point data can also be returned if the source was
+// floating point data (and this is normalized to +/-1.0). The actual number
+// of samples unpacked is returned, which should be equal to the number
+// requested unless the end of fle is encountered or an error occurs.
+
+ulong WavpackUnpackSamples (WavpackContext *wpc, long *buffer, ulong samples)
+{
+    WavpackStream *wps = &wpc->stream;
+    ulong bcount, samples_unpacked = 0, samples_to_unpack;
+    int num_channels = wpc->config.num_channels;
+
+    while (samples) {
+	if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
+	    wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples) {
+		bcount = read_next_header (wpc->infile, &wps->wphdr);
+
+		if (bcount == (ulong) -1)
+		    break;
+
+		if (wps->wphdr.version < 0x402 || wps->wphdr.version > 0x40f) {
+		    strcpy (wpc->error_message, "not compatible with this version of WavPack file!");
+		    break;
+		}
+
+		if (!wps->wphdr.block_samples || wps->sample_index == wps->wphdr.block_index)
+		    if (!unpack_init (wpc))
+			break;
+	}
+
+	if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
+	    wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples)
+		continue;
+
+	if (wps->sample_index < wps->wphdr.block_index) {
+	    samples_to_unpack = wps->wphdr.block_index - wps->sample_index;
+
+	    if (samples_to_unpack > samples)
+		samples_to_unpack = samples;
+
+	    wps->sample_index += samples_to_unpack;
+	    samples_unpacked += samples_to_unpack;
+	    samples -= samples_to_unpack;
+
+	    if (wpc->reduced_channels)
+		samples_to_unpack *= wpc->reduced_channels;
+	    else
+		samples_to_unpack *= num_channels;
+
+	    while (samples_to_unpack--)
+		*buffer++ = 0;
+
+	    continue;
+	}
+
+	samples_to_unpack = wps->wphdr.block_index + wps->wphdr.block_samples - wps->sample_index;
+
+	if (samples_to_unpack > samples)
+	    samples_to_unpack = samples;
+
+	unpack_samples (wpc, buffer, samples_to_unpack);
+
+	if (wpc->reduced_channels)
+	    buffer += samples_to_unpack * wpc->reduced_channels;
+	else
+	    buffer += samples_to_unpack * num_channels;
+
+	samples_unpacked += samples_to_unpack;
+	samples -= samples_to_unpack;
+
+	if (wps->sample_index == wps->wphdr.block_index + wps->wphdr.block_samples) {
+	    if (check_crc_error (wpc))
+		wpc->crc_errors++;
+	}
+
+	if (wps->sample_index == wpc->total_samples)
+	    break;
+    }
+
+    return samples_unpacked;
+}
+
+// Get total number of samples contained in the WavPack file, or -1 if unknown
+
+ulong WavpackGetNumSamples (WavpackContext *wpc)
+{
+    return wpc ? wpc->total_samples : (ulong) -1;
+}
+
+// Get the current sample index position, or -1 if unknown
+
+ulong WavpackGetSampleIndex (WavpackContext *wpc)
+{
+    if (wpc)
+	return wpc->stream.sample_index;
+
+    return (ulong) -1;
+}
+
+// Get the number of errors encountered so far
+
+int WavpackGetNumErrors (WavpackContext *wpc)
+{
+    return wpc ? wpc->crc_errors : 0;
+}
+
+// return TRUE if any uncorrected lossy blocks were actually written or read
+
+int WavpackLossyBlocks (WavpackContext *wpc)
+{
+    return wpc ? wpc->lossy_blocks : 0;
+}
+
+// Returns the sample rate of the specified WavPack file
+
+ulong WavpackGetSampleRate (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.sample_rate : 44100;
+}
+
+// Returns the number of channels of the specified WavPack file. Note that
+// this is the actual number of channels contained in the file, but this
+// version can only decode the first two.
+
+int WavpackGetNumChannels (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.num_channels : 2;
+}
+
+// Returns the actual number of valid bits per sample contained in the
+// original file, which may or may not be a multiple of 8. Floating data
+// always has 32 bits, integers may be from 1 to 32 bits each. When this
+// value is not a multiple of 8, then the "extra" bits are located in the
+// LSBs of the results. That is, values are right justified when unpacked
+// into longs, but are left justified in the number of bytes used by the
+// original data.
+
+int WavpackGetBitsPerSample (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.bits_per_sample : 16;
+}
+
+// Returns the number of bytes used for each sample (1 to 4) in the original
+// file. This is required information for the user of this module because the
+// audio data is returned in the LOWER bytes of the long buffer and must be
+// left-shifted 8, 16, or 24 bits if normalized longs are required.
+
+int WavpackGetBytesPerSample (WavpackContext *wpc)
+{
+    return wpc ? wpc->config.bytes_per_sample : 2;
+}
+
+// This function will return the actual number of channels decoded from the
+// file (which may or may not be less than the actual number of channels, but
+// will always be 1 or 2). Normally, this will be the front left and right
+// channels of a multi-channel file.
+
+int WavpackGetReducedChannels (WavpackContext *wpc)
+{
+    if (wpc)
+	return wpc->reduced_channels ? wpc->reduced_channels : wpc->config.num_channels;
+    else
+	return 2;
+}
+
+// Read from current file position until a valid 32-byte WavPack 4.0 header is
+// found and read into the specified pointer. The number of bytes skipped is
+// returned. If no WavPack header is found within 1 meg, then a -1 is returned
+// to indicate the error. No additional bytes are read past the header and it
+// is returned in the processor's native endian mode. Seeking is not required.
+
+static ulong read_next_header (read_stream infile, WavpackHeader *wphdr)
+{
+    char buffer [sizeof (*wphdr)], *sp = buffer + sizeof (*wphdr), *ep = sp;
+    ulong bytes_skipped = 0;
+    int bleft;
+
+    while (1) {
+	if (sp < ep) {
+	    bleft = ep - sp;
+	    memcpy (buffer, sp, bleft);
+	}
+	else
+	    bleft = 0;
+
+	if (infile (buffer + bleft, sizeof (*wphdr) - bleft) != (long) sizeof (*wphdr) - bleft)
+	    return -1;
+
+	sp = buffer;
+
+	if (*sp++ == 'w' && *sp == 'v' && *++sp == 'p' && *++sp == 'k' &&
+	    !(*++sp & 1) && sp [2] < 16 && !sp [3] && sp [5] == 4 && sp [4] >= 2 && sp [4] <= 0xf) {
+		memcpy (wphdr, buffer, sizeof (*wphdr));
+		little_endian_to_native (wphdr, WavpackHeaderFormat);
+		return bytes_skipped;
+	    }
+
+	while (sp < ep && *sp != 'w')
+	    sp++;
+
+	if ((bytes_skipped += sp - buffer) > 1024 * 1024)
+	    return -1;
+    }
+}