Merge pull request 'WavPack and APEv2 tags support' (#218) from ayumi/tangara-fw:wavpack into main

Reviewed-on: https://codeberg.org/cool-tech-zone/tangara-fw/pulls/218
custom
cooljqln 1 month ago
commit 5995b3a48e
  1. 6
      REUSE.toml
  2. 2
      lib/libtags/CMakeLists.txt
  3. 233
      lib/libtags/ape.c
  4. 4
      lib/libtags/tags.c
  5. 1
      lib/libtags/tags.h
  6. 4
      lib/wavpack/CMakeLists.txt
  7. 141
      lib/wavpack/bits.c
  8. 50
      lib/wavpack/float.c
  9. 25
      lib/wavpack/license.txt
  10. 105
      lib/wavpack/metadata.c
  11. 68
      lib/wavpack/readme.txt
  12. 785
      lib/wavpack/unpack.c
  13. 394
      lib/wavpack/wavpack.h
  14. 560
      lib/wavpack/words.c
  15. 350
      lib/wavpack/wputils.c
  16. 200
      lib/wavpack/wvfilter.c
  17. 4
      src/codecs/CMakeLists.txt
  18. 5
      src/codecs/codec.cpp
  19. 1
      src/codecs/include/types.hpp
  20. 46
      src/codecs/include/wavpack.hpp
  21. 161
      src/codecs/wavpack.cpp
  22. 2
      src/tangara/audio/fatfs_stream_factory.cpp
  23. 3
      src/tangara/database/tag_parser.cpp
  24. 3
      src/tangara/database/tag_parser.hpp
  25. 1
      src/tangara/database/track.hpp
  26. 1
      tools/cmake/common.cmake

@ -181,6 +181,12 @@ precedence = "aggregate"
SPDX-FileCopyrightText = "2002, Xiph.org Foundation" SPDX-FileCopyrightText = "2002, Xiph.org Foundation"
SPDX-License-Identifier = "BSD-3-Clause" SPDX-License-Identifier = "BSD-3-Clause"
[[annotations]]
path = "lib/wavpack/**"
precedence = "aggregate"
SPDX-FileCopyrightText = "1998 - 2006 Conifer Software"
SPDX-License-Identifier = "BSD-3-Clause"
[[annotations]] [[annotations]]
path = "lua/fonts/fusion*" path = "lua/fonts/fusion*"
precedence = "aggregate" precedence = "aggregate"

@ -1,5 +1,5 @@
idf_component_register( idf_component_register(
SRCS 437.c 8859.c flac.c id3genres.c id3v1.c id3v2.c it.c m4a.c mod.c opus.c SRCS 437.c 8859.c ape.c flac.c id3genres.c id3v1.c id3v2.c it.c m4a.c mod.c opus.c
s3m.c tags.c utf16.c vorbis.c wav.c xm.c s3m.c tags.c utf16.c vorbis.c wav.c xm.c
INCLUDE_DIRS . INCLUDE_DIRS .
) )

@ -0,0 +1,233 @@
#include <math.h>
#include "tagspriv.h"
#define leu16int(d) (u16int)(((uchar*)(d))[1]<<8 | ((uchar*)(d))[0]<<0)
enum
{
HeaderSize = 32,
FooterSize = HeaderSize,
MagicOffset = 0,
VersionOffset = 8,
SizeOffset = 12,
CountOffset = 16,
FlagsOffset = 20,
WvHeaderSize = 32,
WvMagicOffset = 0,
WvVersionOffset = 8,
WvSamplesHighOffset = 11,
WvSamplesLowOffset = 12,
WvFlagsOffset = 24,
WvSampleRateMask = 0xf << 23,
WvSampleRateShift = 23,
WvCustomSampleRate = 16,
WvMonoMask = 4,
};
typedef enum
{
TagUTF8,
TagBinary,
TagExternal,
TagReserved,
TagInvalid,
} TagType;
static int
isWavpack(Tagctx *ctx)
{
uchar header[WvHeaderSize];
int size;
u16int version;
u32int flags, samplerate;
uvlong samples;
if(ctx->seek(ctx, 0, 0) < 0)
return 0;
if(ctx->read(ctx, header, WvHeaderSize) != WvHeaderSize)
return 0;
if(memcmp(header+WvMagicOffset, "wvpk", 4))
return 0;
version = leu16int(header+WvVersionOffset);
if(version<0x402 || version>0x410)
return 0;
samples = (uvlong)(*(uchar*)(header+WvSamplesHighOffset))<<32 | leuint(header+WvSamplesLowOffset);
flags = leuint(header+WvFlagsOffset);
if((flags&WvSampleRateMask)>>WvSampleRateShift != WvCustomSampleRate){
const u32int samplerates[] = {6000, 8000, 9600, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000};
samplerate = samplerates[(flags&WvSampleRateMask)>>WvSampleRateShift];
ctx->samplerate = samplerate;
uvlong duration = round((double)samples/samplerate*1000);
ctx->duration = duration;
}
ctx->channels = flags&WvMonoMask ? 1 : 2;
if(ctx->seek(ctx, 0, 0) < 0)
return 0;
if(size = ctx->seek(ctx, 0, 2), size < 0)
return 0;
ctx->bitrate = (double)size*8.0/(samples/samplerate)/1000;
return 1;
}
static int
detectFormat(Tagctx *ctx)
{
if(isWavpack(ctx))
return Fwavpack;
return Funknown;
}
static int
tagHasHeader(u32int tags)
{
return (tags&(1<<31)) >> 31;
}
static int
tagIsHeader(u32int tags)
{
return (tags&(1<<29)) >> 29;
}
static TagType
tagGetType(u32int tags)
{
switch((tags&(1<<1))>>1 | (tags&(1<<2))>>1 | (tags&(1<<3))>>1 | (tags&(1<<4))>>1){
case 0:
return TagUTF8;
case 1:
return TagBinary;
case 2:
return TagExternal;
case 3:
return TagReserved;
default:
return TagInvalid;
}
}
static int
tagTagType(char *name)
{
if(!strcmp(name, "Album"))
return Talbum;
else if(!strcmp(name, "Album Artist"))
return Talbumartist;
else if(!strcmp(name, "Artist"))
return Tartist;
else if(!strcmp(name, "Comment"))
return Tcomment;
else if(!strcmp(name, "Composer"))
return Tcomposer;
else if(!strncmp(name, "Cover Art (", 11)){
if(name[strlen(name)-1] == ')')
return Timage;
}else if(!strcmp(name, "Genre"))
return Tgenre;
else if(!strcmp(name, "Replaygain_Album_Gain"))
return Talbumgain;
else if(!strcmp(name, "Replaygain_Album_Peak"))
return Talbumpeak;
else if(!strcmp(name, "Replaygain_Track_Gain"))
return Ttrackgain;
else if(!strcmp(name, "Replaygain_Track_Peak"))
return Ttrackpeak;
else if(!strcmp(name, "Title"))
return Ttitle;
else if(!strcmp(name, "Track"))
return Ttrack;
else if(!strcmp(name, "Year"))
return Tdate;
return Tunknown;
}
int
tagape(Tagctx *ctx)
{
uchar footer[FooterSize];
u32int i, count;
ctx->format = detectFormat(ctx);
if(ctx->seek(ctx, -FooterSize, 2) < 0)
return -1;
if(ctx->read(ctx, footer, FooterSize) != FooterSize)
return -1;
if(memcmp(footer+MagicOffset, "APETAGEX", 8))
return -1;
if(leuint(footer+VersionOffset) != 2000)
return -1;
if(tagIsHeader(leuint(footer+FlagsOffset)))
return -1;
if(ctx->seek(ctx, -FooterSize-leuint(footer+SizeOffset), 2) < 0)
return -1;
if(tagHasHeader(leuint(footer+FlagsOffset))){
uchar header[HeaderSize];
if(ctx->read(ctx, header, HeaderSize) != HeaderSize)
return -1;
if(memcmp(header, footer, 23))
return -1;
if(!tagHasHeader(leuint(header+FlagsOffset)))
return -1;
if(!tagIsHeader(leuint(header+FlagsOffset)))
return -1;
}else if(ctx->seek(ctx, HeaderSize, 1) < 0)
return -1;
for(i = 0, count = leuint(footer+CountOffset); i < count; i++){
int valueOffset = 0;
char c;
u32int d, length, flags;
if(ctx->read(ctx, &d, 4) != 4)
return -1;
length = leuint(&d);
if(ctx->read(ctx, &d, 4) != 4)
return -1;
flags = leuint(&d);
do{
if(valueOffset == ctx->bufsz)
return -1;
if(ctx->read(ctx, &c, 1) != 1)
return -1;
if(c<' ' || c>'~')
if(c != '\0')
return -1;
ctx->buf[valueOffset++] = c;
}while(c != '\0');
if(valueOffset+1+(int)length>ctx->bufsz && tagTagType(ctx->buf)!=Timage){
if(ctx->seek(ctx, length, 1) < 0)
return -1;
continue;
}
switch(tagGetType(flags)){
u32int keyOffset;
case TagUTF8:
if(ctx->read(ctx, ctx->buf+valueOffset, length) != (int)length)
return -1;
(ctx->buf+valueOffset)[length] = '\0';
for(keyOffset = 0; keyOffset != length;){
txtcb(ctx, tagTagType(ctx->buf), ctx->buf, ctx->buf+valueOffset+keyOffset);
if(keyOffset += strlen(ctx->buf+valueOffset+keyOffset), keyOffset != length)
keyOffset++;
}
break;
case TagBinary:
if(tagTagType(ctx->buf) == Timage)
tagscallcb(ctx, Timage, ctx->buf, ctx->buf, ctx->seek(ctx, 0, 1), length, NULL);
if(ctx->seek(ctx, length, 1) < 0)
return -1;
break;
default:
if(ctx->seek(ctx, length, 1) < 0)
return -1;
}
}
return 0;
}

@ -8,6 +8,7 @@ struct Getter
int format; int format;
}; };
extern int tagape(Tagctx *ctx);
extern int tagflac(Tagctx *ctx); extern int tagflac(Tagctx *ctx);
extern int tagid3v1(Tagctx *ctx); extern int tagid3v1(Tagctx *ctx);
extern int tagid3v2(Tagctx *ctx); extern int tagid3v2(Tagctx *ctx);
@ -22,6 +23,7 @@ extern int tagmod(Tagctx *ctx);
static const Getter g[] = static const Getter g[] =
{ {
{tagape, Funknown},
{tagid3v2, Fmp3}, {tagid3v2, Fmp3},
{tagid3v1, Fmp3}, {tagid3v1, Fmp3},
{tagvorbis, Fogg}, {tagvorbis, Fogg},
@ -71,7 +73,9 @@ tagsget(Tagctx *ctx)
for(i = 0; i < nelem(g); i++){ for(i = 0; i < nelem(g); i++){
ctx->num = 0; ctx->num = 0;
if(g[i].f(ctx) == 0){ if(g[i].f(ctx) == 0){
if(ctx->format == Funknown){
ctx->format = g[i].format; ctx->format = g[i].format;
}
res = 0; res = 0;
} }
ctx->seek(ctx, ctx->restart, 0); ctx->seek(ctx, ctx->restart, 0);

@ -37,6 +37,7 @@ enum
Fm4a, Fm4a,
Fopus, Fopus,
Fwav, Fwav,
Fwavpack,
Fit, Fit,
Fxm, Fxm,
Fs3m, Fs3m,

@ -0,0 +1,4 @@
idf_component_register(
SRCS bits.c float.c wputils.c metadata.c unpack.c words.c
INCLUDE_DIRS .
)

@ -0,0 +1,141 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 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, void *user_data, uint32_t 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;
bs->user_data = user_data;
}
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) {
uint32_t 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->user_data, 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;
int32_t temp;
while (*format) {
switch (*format) {
case 'L':
temp = cp [0] + ((int32_t) cp [1] << 8) + ((int32_t) cp [2] << 16) + ((int32_t) cp [3] << 24);
* (int32_t *) cp = temp;
cp += 4;
break;
case 'S':
temp = cp [0] + (cp [1] << 8);
* (short *) cp = (short) temp;
cp += 2;
break;
default:
if (isdigit ((unsigned char) *format))
cp += *format - '0';
break;
}
format++;
}
}
void native_to_little_endian (void *data, char *format)
{
uchar *cp = (uchar *) data;
int32_t temp;
while (*format) {
switch (*format) {
case 'L':
temp = * (int32_t *) 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 ((unsigned char) *format))
cp += *format - '0';
break;
}
format++;
}
}

@ -0,0 +1,50 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 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, int32_t *values, int32_t num_values)
{
int shift = wps->float_max_exp - wps->float_norm_exp + wps->float_shift;
if (shift > 32)
shift = 32;
else if (shift < -32)
shift = -32;
while (num_values--) {
if (shift > 0)
*values <<= shift;
else if (shift < 0)
*values >>= -shift;
if (*values > 8388607L)
*values = 8388607L;
else if (*values < -8388608L)
*values = -8388608L;
values++;
}
}

@ -0,0 +1,25 @@
Copyright (c) 1998 - 2006 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.

@ -0,0 +1,105 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 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 (wpc->user_data, &wpmd->id, 1) || !wpc->infile (wpc->user_data, &tchar, 1))
return FALSE;
wpmd->byte_length = tchar << 1;
if (wpmd->id & ID_LARGE) {
wpmd->id &= ~ID_LARGE;
if (!wpc->infile (wpc->user_data, &tchar, 1))
return FALSE;
wpmd->byte_length += (int32_t) tchar << 9;
if (!wpc->infile (wpc->user_data, &tchar, 1))
return FALSE;
wpmd->byte_length += (int32_t) 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)) {
uint32_t bytes_to_read = wpmd->byte_length + (wpmd->byte_length & 1);
if (wpc->infile (wpc->user_data, wpc->read_buffer, bytes_to_read) != (int32_t) 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;
}
}

@ -0,0 +1,68 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 Conifer Software. //
// All Rights Reserved. //
// Distributed under the BSD Software License (see license.txt) //
////////////////////////////////////////////////////////////////////////////
This package contains a tiny version of the WavPack 4.40 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. The WIN32 macro
is used for Windows to force the stdin and stdout streams to be binary mode.
Compared to the previous version, this library has been optimized somewhat
for improved performance in exchange for slightly larger code size. The
library also now includes hand-optimized assembly language versions of the
decorrelation functions for both the ColdFire (w/EMAC) and ARM processors.
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.
The previous version of this library would handle float files by returning
32-bit floating-point data (even though no floating point math was used).
Because this library would normally be used for simply playing WavPack
files where lossless performance (beyond 24-bits) is not relevant, I have
changed this behavior. Now, these files will generate clipped 24-bit data.
The MODE_FLOAT flag will still be returned by WavpackGetMode(), but the
BitsPerSample and BytesPerSample queries will be 24 and 3, respectfully.
What this means is that an application that can handle 24-bit data will
now be able to handle floating point data (assuming that the MODE_FLOAT
flag is ignored).
To make this code viable on the greatest number of hardware platforms, the
following are true:
speed is about 5x 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 40K
no malloc / free usage
To maintain compatibility on various platforms, the following conventions
are used:
a "char" must be exactly 8-bits
a "short" must be exactly 16-bits
an "int" must be at least 16-bits, but may be larger
the "long" type is not used to avoid problems with 64-bit compilers
Questions or comments should be directed to david@wavpack.com

@ -0,0 +1,785 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 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 <stdlib.h>
#include <string.h>
#define LOSSY_MUTE
///////////////////////////// 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 != (uint32_t) -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, (unsigned char *) wpmd->data + wpmd->byte_length, NULL, NULL, 0);
else if (wpmd->byte_length)
bs_open_read (&wps->wvbits, wpc->read_buffer, wpc->read_buffer + sizeof (wpc->read_buffer),
wpc->infile, wpc->user_data, 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;
signed char *byteptr = wpmd->data;
struct decorr_pass *dpp;
if (!(wps->wphdr.flags & MONO_DATA))
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_DATA))
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_DATA))
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_DATA)) {
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_DATA)) {
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;
uint32_t mask = 0;
if (!bytecnt || bytecnt > 5)
return FALSE;
wpc->config.num_channels = *byteptr++;
while (--bytecnt) {
mask |= (uint32_t) *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 |= (int32_t) *byteptr++ << 8;
wpc->config.flags |= (int32_t) *byteptr++ << 16;
wpc->config.flags |= (int32_t) *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 int32_ts. 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.
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
extern void decorr_stereo_pass_cont_mcf5249 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
#elif defined(CPU_ARM) && !defined(SIMULATOR)
extern void decorr_stereo_pass_cont_arm (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
extern void decorr_stereo_pass_cont_arml (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
#else
static void decorr_stereo_pass_cont (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
#endif
static void decorr_mono_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void decorr_stereo_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void fixup_samples (WavpackStream *wps, int32_t *buffer, uint32_t sample_count);
int32_t unpack_samples (WavpackContext *wpc, int32_t *buffer, uint32_t sample_count)
{
WavpackStream *wps = &wpc->stream;
uint32_t flags = wps->wphdr.flags, crc = wps->crc, i;
int32_t mute_limit = (1L << ((flags & MAG_MASK) >> MAG_LSB)) + 2;
struct decorr_pass *dpp;
int32_t *bptr, *eptr;
int tcount;
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_DATA) {
eptr = buffer + sample_count;
i = get_words (buffer, sample_count, flags, &wps->w, &wps->wvbits);
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
decorr_mono_pass (dpp, buffer, sample_count);
for (bptr = buffer; bptr < eptr; ++bptr) {
if (labs (bptr [0]) > mute_limit) {
i = bptr - buffer;
break;
}
crc = crc * 3 + bptr [0];
}
}
//////////////////// handle version 4 stereo data ////////////////////////
else {
eptr = buffer + (sample_count * 2);
i = get_words (buffer, sample_count, flags, &wps->w, &wps->wvbits);
if (sample_count < 16)
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
decorr_stereo_pass (dpp, buffer, sample_count);
else
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
decorr_stereo_pass (dpp, buffer, 8);
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
decorr_stereo_pass_cont_mcf5249 (dpp, buffer + 16, sample_count - 8);
#elif defined(CPU_ARM) && !defined(SIMULATOR)
if (((flags & MAG_MASK) >> MAG_LSB) > 15)
decorr_stereo_pass_cont_arml (dpp, buffer + 16, sample_count - 8);
else
decorr_stereo_pass_cont_arm (dpp, buffer + 16, sample_count - 8);
#else
decorr_stereo_pass_cont (dpp, buffer + 16, sample_count - 8);
#endif
}
if (flags & JOINT_STEREO)
for (bptr = buffer; bptr < eptr; bptr += 2) {
bptr [0] += (bptr [1] -= (bptr [0] >> 1));
if (labs (bptr [0]) > mute_limit || labs (bptr [1]) > mute_limit) {
i = (bptr - buffer) / 2;
break;
}
crc = (crc * 3 + bptr [0]) * 3 + bptr [1];
}
else
for (bptr = buffer; bptr < eptr; bptr += 2) {
if (labs (bptr [0]) > mute_limit || labs (bptr [1]) > mute_limit) {
i = (bptr - buffer) / 2;
break;
}
crc = (crc * 3 + bptr [0]) * 3 + bptr [1];
}
}
if (i != sample_count) {
memset (buffer, 0, sample_count * (flags & MONO_FLAG ? 4 : 8));
wps->mute_error = TRUE;
i = sample_count;
}
fixup_samples (wps, buffer, i);
if (flags & FALSE_STEREO) {
int32_t *dptr = buffer + i * 2;
int32_t *sptr = buffer + i;
int32_t c = i;
while (c--) {
*--dptr = *--sptr;
*--dptr = *sptr;
}
}
wps->sample_index += i;
wps->crc = crc;
return i;
}
static void decorr_stereo_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t delta = dpp->delta, weight_A = dpp->weight_A, weight_B = dpp->weight_B;
int32_t *bptr, *eptr = buffer + (sample_count * 2), sam_A, sam_B;
int m, k;
switch (dpp->term) {
case 17:
for (bptr = buffer; bptr < eptr; bptr += 2) {
sam_A = 2 * dpp->samples_A [0] - dpp->samples_A [1];
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
update_weight (weight_A, delta, sam_A, bptr [0]);
bptr [0] = dpp->samples_A [0];
sam_A = 2 * dpp->samples_B [0] - dpp->samples_B [1];
dpp->samples_B [1] = dpp->samples_B [0];
dpp->samples_B [0] = apply_weight (weight_B, sam_A) + bptr [1];
update_weight (weight_B, delta, sam_A, bptr [1]);
bptr [1] = dpp->samples_B [0];
}
break;
case 18:
for (bptr = buffer; bptr < eptr; bptr += 2) {
sam_A = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
update_weight (weight_A, delta, sam_A, bptr [0]);
bptr [0] = dpp->samples_A [0];
sam_A = (3 * dpp->samples_B [0] - dpp->samples_B [1]) >> 1;
dpp->samples_B [1] = dpp->samples_B [0];
dpp->samples_B [0] = apply_weight (weight_B, sam_A) + bptr [1];
update_weight (weight_B, delta, sam_A, bptr [1]);
bptr [1] = dpp->samples_B [0];
}
break;
default:
for (m = 0, k = dpp->term & (MAX_TERM - 1), bptr = buffer; bptr < eptr; bptr += 2) {
sam_A = dpp->samples_A [m];
dpp->samples_A [k] = apply_weight (weight_A, sam_A) + bptr [0];
update_weight (weight_A, delta, sam_A, bptr [0]);
bptr [0] = dpp->samples_A [k];
sam_A = dpp->samples_B [m];
dpp->samples_B [k] = apply_weight (weight_B, sam_A) + bptr [1];
update_weight (weight_B, delta, sam_A, bptr [1]);
bptr [1] = dpp->samples_B [k];
m = (m + 1) & (MAX_TERM - 1);
k = (k + 1) & (MAX_TERM - 1);
}
if (m) {
int32_t temp_samples [MAX_TERM];
memcpy (temp_samples, dpp->samples_A, sizeof (dpp->samples_A));
for (k = 0; k < MAX_TERM; k++, m++)
dpp->samples_A [k] = temp_samples [m & (MAX_TERM - 1)];
memcpy (temp_samples, dpp->samples_B, sizeof (dpp->samples_B));
for (k = 0; k < MAX_TERM; k++, m++)
dpp->samples_B [k] = temp_samples [m & (MAX_TERM - 1)];
}
break;
case -1:
for (bptr = buffer; bptr < eptr; bptr += 2) {
sam_A = bptr [0] + apply_weight (weight_A, dpp->samples_A [0]);
update_weight_clip (weight_A, delta, dpp->samples_A [0], bptr [0]);
bptr [0] = sam_A;
dpp->samples_A [0] = bptr [1] + apply_weight (weight_B, sam_A);
update_weight_clip (weight_B, delta, sam_A, bptr [1]);
bptr [1] = dpp->samples_A [0];
}
break;
case -2:
for (bptr = buffer; bptr < eptr; bptr += 2) {
sam_B = bptr [1] + apply_weight (weight_B, dpp->samples_B [0]);
update_weight_clip (weight_B, delta, dpp->samples_B [0], bptr [1]);
bptr [1] = sam_B;
dpp->samples_B [0] = bptr [0] + apply_weight (weight_A, sam_B);
update_weight_clip (weight_A, delta, sam_B, bptr [0]);
bptr [0] = dpp->samples_B [0];
}
break;
case -3:
for (bptr = buffer; bptr < eptr; bptr += 2) {
sam_A = bptr [0] + apply_weight (weight_A, dpp->samples_A [0]);
update_weight_clip (weight_A, delta, dpp->samples_A [0], bptr [0]);
sam_B = bptr [1] + apply_weight (weight_B, dpp->samples_B [0]);
update_weight_clip (weight_B, delta, dpp->samples_B [0], bptr [1]);
bptr [0] = dpp->samples_B [0] = sam_A;
bptr [1] = dpp->samples_A [0] = sam_B;
}
break;
}
dpp->weight_A = weight_A;
dpp->weight_B = weight_B;
}
#if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM)) || defined(SIMULATOR)
static void decorr_stereo_pass_cont (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t delta = dpp->delta, weight_A = dpp->weight_A, weight_B = dpp->weight_B;
int32_t *bptr, *tptr, *eptr = buffer + (sample_count * 2), sam_A, sam_B;
int k, i;
switch (dpp->term) {
case 17:
for (bptr = buffer; bptr < eptr; bptr += 2) {
sam_A = 2 * bptr [-2] - bptr [-4];
bptr [0] = apply_weight (weight_A, sam_A) + (sam_B = bptr [0]);
update_weight (weight_A, delta, sam_A, sam_B);
sam_A = 2 * bptr [-1] - bptr [-3];
bptr [1] = apply_weight (weight_B, sam_A) + (sam_B = bptr [1]);
update_weight (weight_B, delta, sam_A, sam_B);
}
dpp->samples_B [0] = bptr [-1];
dpp->samples_A [0] = bptr [-2];
dpp->samples_B [1] = bptr [-3];
dpp->samples_A [1] = bptr [-4];
break;
case 18:
for (bptr = buffer; bptr < eptr; bptr += 2) {
sam_A = (3 * bptr [-2] - bptr [-4]) >> 1;
bptr [0] = apply_weight (weight_A, sam_A) + (sam_B = bptr [0]);
update_weight (weight_A, delta, sam_A, sam_B);
sam_A = (3 * bptr [-1] - bptr [-3]) >> 1;
bptr [1] = apply_weight (weight_B, sam_A) + (sam_B = bptr [1]);
update_weight (weight_B, delta, sam_A, sam_B);
}
dpp->samples_B [0] = bptr [-1];
dpp->samples_A [0] = bptr [-2];
dpp->samples_B [1] = bptr [-3];
dpp->samples_A [1] = bptr [-4];
break;
default:
for (bptr = buffer, tptr = buffer - (dpp->term * 2); bptr < eptr; bptr += 2, tptr += 2) {
bptr [0] = apply_weight (weight_A, tptr [0]) + (sam_A = bptr [0]);
update_weight (weight_A, delta, tptr [0], sam_A);
bptr [1] = apply_weight (weight_B, tptr [1]) + (sam_A = bptr [1]);
update_weight (weight_B, delta, tptr [1], sam_A);
}
for (k = dpp->term - 1, i = 8; i--; k--) {
dpp->samples_B [k & (MAX_TERM - 1)] = *--bptr;
dpp->samples_A [k & (MAX_TERM - 1)] = *--bptr;
}
break;
case -1:
for (bptr = buffer; bptr < eptr; bptr += 2) {
bptr [0] = apply_weight (weight_A, bptr [-1]) + (sam_A = bptr [0]);
update_weight_clip (weight_A, delta, bptr [-1], sam_A);
bptr [1] = apply_weight (weight_B, bptr [0]) + (sam_A = bptr [1]);
update_weight_clip (weight_B, delta, bptr [0], sam_A);
}
dpp->samples_A [0] = bptr [-1];
break;
case -2:
for (bptr = buffer; bptr < eptr; bptr += 2) {
bptr [1] = apply_weight (weight_B, bptr [-2]) + (sam_A = bptr [1]);
update_weight_clip (weight_B, delta, bptr [-2], sam_A);
bptr [0] = apply_weight (weight_A, bptr [1]) + (sam_A = bptr [0]);
update_weight_clip (weight_A, delta, bptr [1], sam_A);
}
dpp->samples_B [0] = bptr [-2];
break;
case -3:
for (bptr = buffer; bptr < eptr; bptr += 2) {
bptr [0] = apply_weight (weight_A, bptr [-1]) + (sam_A = bptr [0]);
update_weight_clip (weight_A, delta, bptr [-1], sam_A);
bptr [1] = apply_weight (weight_B, bptr [-2]) + (sam_A = bptr [1]);
update_weight_clip (weight_B, delta, bptr [-2], sam_A);
}
dpp->samples_A [0] = bptr [-1];
dpp->samples_B [0] = bptr [-2];
break;
}
dpp->weight_A = weight_A;
dpp->weight_B = weight_B;
}
#endif
static void decorr_mono_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t delta = dpp->delta, weight_A = dpp->weight_A;
int32_t *bptr, *eptr = buffer + sample_count, sam_A;
int m, k;
switch (dpp->term) {
case 17:
for (bptr = buffer; bptr < eptr; bptr++) {
sam_A = 2 * dpp->samples_A [0] - dpp->samples_A [1];
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
update_weight (weight_A, delta, sam_A, bptr [0]);
bptr [0] = dpp->samples_A [0];
}
break;
case 18:
for (bptr = buffer; bptr < eptr; bptr++) {
sam_A = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
update_weight (weight_A, delta, sam_A, bptr [0]);
bptr [0] = dpp->samples_A [0];
}
break;
default:
for (m = 0, k = dpp->term & (MAX_TERM - 1), bptr = buffer; bptr < eptr; bptr++) {
sam_A = dpp->samples_A [m];
dpp->samples_A [k] = apply_weight (weight_A, sam_A) + bptr [0];
update_weight (weight_A, delta, sam_A, bptr [0]);
bptr [0] = dpp->samples_A [k];
m = (m + 1) & (MAX_TERM - 1);
k = (k + 1) & (MAX_TERM - 1);
}
if (m) {
int32_t temp_samples [MAX_TERM];
memcpy (temp_samples, dpp->samples_A, sizeof (dpp->samples_A));
for (k = 0; k < MAX_TERM; k++, m++)
dpp->samples_A [k] = temp_samples [m & (MAX_TERM - 1)];
}
break;
}
dpp->weight_A = weight_A;
}
// 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, int32_t *buffer, uint32_t sample_count)
{
uint32_t 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) {
uint32_t 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;
int32_t *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) {
int32_t 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:
default:
min_shifted = (min_value = (int32_t) 0x80000000 >> shift) << shift;
max_shifted = (max_value = (int32_t) 0x7FFFFFFF >> 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;
}

@ -0,0 +1,394 @@
////////////////////////////////////////////////////////////////////////////
// **** 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
#ifdef __cplusplus
extern "C" {
#endif
#include <sys/types.h>
// This header file contains all the definitions required by WavPack.
#ifdef __BORLANDC__
typedef unsigned long uint32_t;
typedef long int32_t;
#elif defined(_WIN32) && !defined(__MINGW32__)
#include <stdlib.h>
typedef unsigned __int64 uint64_t;
typedef unsigned __int32 uint32_t;
typedef __int64 int64_t;
typedef __int32 int32_t;
#else
#include <inttypes.h>
#endif
typedef unsigned char uchar;
#if !defined(__GNUC__) || defined(WIN32)
typedef unsigned short ushort;
typedef unsigned int uint;
#endif
#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];
uint32_t ckSize;
short version;
uchar track_no, index_no;
uint32_t 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 FALSE_STEREO 0x40000000 // block is stereo, but data is mono
#define IGNORED_FLAGS 0x18000000 // reserved, but ignore if encountered
#define NEW_SHAPING 0x20000000 // use IIR filter for negative shaping
#define UNKNOWN_FLAGS 0x80000000 // also reserved, but refuse decode if
// encountered
#define MONO_DATA (MONO_FLAG | FALSE_STEREO)
#define MIN_STREAM_VERS 0x402 // lowest stream version we'll decode
#define MAX_STREAM_VERS 0x410 // highest stream version we'll decode
//////////////////////////// WavPack Metadata /////////////////////////////////
// This is an internal representation of metadata.
typedef struct {
int32_t 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 num_channels, float_norm_exp;
uint32_t 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_FAST_FLAG 0x200 // fast mode
#define CONFIG_HIGH_FLAG 0x800 // high quality mode
#define CONFIG_VERY_HIGH_FLAG 0x1000 // very high
#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_CREATE_EXE 0x40000 // create executable
#define CONFIG_CREATE_WVC 0x80000 // create correction file
#define CONFIG_OPTIMIZE_WVC 0x100000 // maximize bybrid compression
#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_OPTIMIZE_MONO 0x80000000 // optimize for mono streams posing as stereo
//////////////////////////////// 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 int32_t (*read_stream)(void *, void *, int32_t);
typedef struct bs {
uchar *buf, *end, *ptr;
void (*wrap)(struct bs *bs);
uint32_t file_bytes, sr;
int error, bc;
read_stream file;
void *user_data;
} Bitstream;
#define MAX_NTERMS 16
#define MAX_TERM 8
struct decorr_pass {
short term, delta, weight_A, weight_B;
int32_t samples_A [MAX_TERM], samples_B [MAX_TERM];
};
struct entropy_data {
uint32_t median [3], slow_level, error_limit;
};
struct words_data {
uint32_t bitrate_delta [2], bitrate_acc [2];
uint32_t pend_data, holding_one, zeros_acc;
int holding_zero, pend_count;
struct entropy_data c [2];
};
typedef struct {
WavpackHeader wphdr;
Bitstream wvbits;
struct words_data w;
int num_terms, mute_error;
uint32_t sample_index, crc;
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];
} 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;
void *user_data;
uint32_t 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, void *user_data, uint32_t 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 |= (int32_t)*((bs)->ptr) << (bs)->bc; \
(bs)->bc += 8; \
} \
*(value) = (bs)->sr; \
if ((bs)->bc > 32) { \
(bs)->bc -= (nbits); \
(bs)->sr = *((bs)->ptr) >> (8 - (bs)->bc); \
} \
else { \
(bs)->bc -= (nbits); \
(bs)->sr >>= (nbits); \
} \
}
void little_endian_to_native (void *data, char *format);
void native_to_little_endian (void *data, char *format);
// These macros implement the weight application and update operations
// that are at the heart of the decorrelation loops. Note that when there
// are several alternative versions of the same macro (marked with PERFCOND)
// then the versions are functionally equivalent with respect to WavPack
// decoding and the user should choose the one that provides the best
// performance. This may be easier to check when NOT using the assembly
// language optimizations.
#if 1 // PERFCOND
#define apply_weight_i(weight, sample) ((weight * sample + 512) >> 10)
#else
#define apply_weight_i(weight, sample) ((((weight * sample) >> 8) + 2) >> 2)
#endif
#define apply_weight_f(weight, sample) (((((sample & 0xffffL) * weight) >> 9) + \
(((sample & ~0xffffL) >> 9) * weight) + 1) >> 1)
#if 0 // PERFCOND
#define apply_weight(weight, sample) (sample != (short) sample ? \
apply_weight_f (weight, sample) : apply_weight_i (weight, sample))
#else
#define apply_weight(weight, sample) ((int32_t)((weight * (int64_t) sample + 512) >> 10))
#endif
#if 0 // PERFCOND
#define update_weight(weight, delta, source, result) \
if (source && result) { int32_t s = (int32_t) (source ^ result) >> 31; weight = (delta ^ s) + (weight - s); }
#elif 0
#define update_weight(weight, delta, source, result) \
if (source && result) weight += (((source ^ result) >> 30) | 1) * delta
#else
#define update_weight(weight, delta, source, result) \
if (source && result) (source ^ result) < 0 ? (weight -= delta) : (weight += delta)
#endif
#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
// 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);
int32_t unpack_samples (WavpackContext *wpc, int32_t *buffer, uint32_t 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);
int32_t get_words (int32_t *buffer, int nsamples, uint32_t flags,
struct words_data *w, Bitstream *bs);
int32_t exp2s (int log);
int restore_weight (signed char weight);
#define WORD_EOF (1L << 31)
// float.c
int read_float_info (WavpackStream *wps, WavpackMetadata *wpmd);
void float_values (WavpackStream *wps, int32_t *values, int32_t num_values);
// wputils.c
int WavpackOpenFileInput (WavpackContext *wpc, read_stream infile, void *user_data, 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
uint32_t WavpackUnpackSamples (WavpackContext *wpc, int32_t *buffer, uint32_t samples);
uint32_t WavpackGetNumSamples (WavpackContext *wpc);
uint32_t WavpackGetSampleIndex (WavpackContext *wpc);
int WavpackGetNumErrors (WavpackContext *wpc);
int WavpackLossyBlocks (WavpackContext *wpc);
uint32_t WavpackGetSampleRate (WavpackContext *wpc);
int WavpackGetBitsPerSample (WavpackContext *wpc);
int WavpackGetBytesPerSample (WavpackContext *wpc);
int WavpackGetNumChannels (WavpackContext *wpc);
int WavpackGetReducedChannels (WavpackContext *wpc);
#ifdef __cplusplus
}
#endif

@ -0,0 +1,560 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 Conifer Software. //
// All Rights Reserved. //
// Distributed under the BSD Software License (see license.txt) //
////////////////////////////////////////////////////////////////////////////
// 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) (((c->median [med]) >> 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() (c->median [0] += ((c->median [0] + DIV0) / DIV0) * 5)
#define DEC_MED0() (c->median [0] -= ((c->median [0] + (DIV0-2)) / DIV0) * 2)
#define INC_MED1() (c->median [1] += ((c->median [1] + DIV1) / DIV1) * 5)
#define DEC_MED1() (c->median [1] -= ((c->median [1] + (DIV1-2)) / DIV1) * 2)
#define INC_MED2() (c->median [2] += ((c->median [2] + DIV2) / DIV2) * 5)
#define DEC_MED2() (c->median [2] -= ((c->median [2] + (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
};
static const char ones_count_table [] = {
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,6,
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,7,
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,6,
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,8
};
///////////////////////////// executable code ////////////////////////////////
void init_words (WavpackStream *wps)
{
CLEAR (wps->w);
}
static int mylog2 (uint32_t 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_DATA) ? 6 : 12))
return FALSE;
wps->w.c [0].median [0] = exp2s (byteptr [0] + (byteptr [1] << 8));
wps->w.c [0].median [1] = exp2s (byteptr [2] + (byteptr [3] << 8));
wps->w.c [0].median [2] = exp2s (byteptr [4] + (byteptr [5] << 8));
if (!(wps->wphdr.flags & MONO_DATA)) {
wps->w.c [1].median [0] = exp2s (byteptr [6] + (byteptr [7] << 8));
wps->w.c [1].median [1] = exp2s (byteptr [8] + (byteptr [9] << 8));
wps->w.c [1].median [2] = 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.c [0].slow_level = exp2s (byteptr [0] + (byteptr [1] << 8));
byteptr += 2;
if (!(wps->wphdr.flags & MONO_DATA)) {
wps->w.c [1].slow_level = exp2s (byteptr [0] + (byteptr [1] << 8));
byteptr += 2;
}
}
wps->w.bitrate_acc [0] = (int32_t)(byteptr [0] + (byteptr [1] << 8)) << 16;
byteptr += 2;
if (!(wps->wphdr.flags & MONO_DATA)) {
wps->w.bitrate_acc [1] = (int32_t)(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_DATA)) {
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.
void update_error_limit (struct words_data *w, uint32_t flags)
{
int bitrate_0 = (w->bitrate_acc [0] += w->bitrate_delta [0]) >> 16;
if (flags & MONO_DATA) {
if (flags & HYBRID_BITRATE) {
int slow_log_0 = (w->c [0].slow_level + SLO) >> SLS;
if (slow_log_0 - bitrate_0 > -0x100)
w->c [0].error_limit = exp2s (slow_log_0 - bitrate_0 + 0x100);
else
w->c [0].error_limit = 0;
}
else
w->c [0].error_limit = exp2s (bitrate_0);
}
else {
int bitrate_1 = (w->bitrate_acc [1] += w->bitrate_delta [1]) >> 16;
if (flags & HYBRID_BITRATE) {
int slow_log_0 = (w->c [0].slow_level + SLO) >> SLS;
int slow_log_1 = (w->c [1].slow_level + SLO) >> SLS;
if (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)
w->c [0].error_limit = exp2s (slow_log_0 - bitrate_0 + 0x100);
else
w->c [0].error_limit = 0;
if (slow_log_1 - bitrate_1 > -0x100)
w->c [1].error_limit = exp2s (slow_log_1 - bitrate_1 + 0x100);
else
w->c [1].error_limit = 0;
}
else {
w->c [0].error_limit = exp2s (bitrate_0);
w->c [1].error_limit = exp2s (bitrate_1);
}
}
}
static uint32_t read_code (Bitstream *bs, uint32_t 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.
int32_t get_words (int32_t *buffer, int nsamples, uint32_t flags,
struct words_data *w, Bitstream *bs)
{
register struct entropy_data *c = w->c;
int csamples;
if (!(flags & MONO_DATA))
nsamples *= 2;
for (csamples = 0; csamples < nsamples; ++csamples) {
uint32_t ones_count, low, mid, high;
if (!(flags & MONO_DATA))
c = w->c + (csamples & 1);
if (!(w->c [0].median [0] & ~1) && !w->holding_zero && !w->holding_one && !(w->c [1].median [0] & ~1)) {
uint32_t mask;
int cbits;
if (w->zeros_acc) {
if (--w->zeros_acc) {
c->slow_level -= (c->slow_level + SLO) >> SLS;
*buffer++ = 0;
continue;
}
}
else {
for (cbits = 0; cbits < 33 && getbit (bs); ++cbits);
if (cbits == 33)
break;
if (cbits < 2)
w->zeros_acc = cbits;
else {
for (mask = 1, w->zeros_acc = 0; --cbits; mask <<= 1)
if (getbit (bs))
w->zeros_acc |= mask;
w->zeros_acc |= mask;
}
if (w->zeros_acc) {
c->slow_level -= (c->slow_level + SLO) >> SLS;
CLEAR (w->c [0].median);
CLEAR (w->c [1].median);
*buffer++ = 0;
continue;
}
}
}
if (w->holding_zero)
ones_count = w->holding_zero = 0;
else {
int next8;
if (bs->bc < 8) {
if (++(bs->ptr) == bs->end)
bs->wrap (bs);
next8 = (bs->sr |= *(bs->ptr) << bs->bc) & 0xff;
bs->bc += 8;
}
else
next8 = bs->sr & 0xff;
if (next8 == 0xff) {
bs->bc -= 8;
bs->sr >>= 8;
for (ones_count = 8; ones_count < (LIMIT_ONES + 1) && getbit (bs); ++ones_count);
if (ones_count == (LIMIT_ONES + 1))
break;
if (ones_count == LIMIT_ONES) {
uint32_t mask;
int cbits;
for (cbits = 0; cbits < 33 && getbit (bs); ++cbits);
if (cbits == 33)
break;
if (cbits < 2)
ones_count = cbits;
else {
for (mask = 1, ones_count = 0; --cbits; mask <<= 1)
if (getbit (bs))
ones_count |= mask;
ones_count |= mask;
}
ones_count += LIMIT_ONES;
}
}
else {
bs->bc -= (ones_count = ones_count_table [next8]) + 1;
bs->sr >>= ones_count + 1;
}
if (w->holding_one) {
w->holding_one = ones_count & 1;
ones_count = (ones_count >> 1) + 1;
}
else {
w->holding_one = ones_count & 1;
ones_count >>= 1;
}
w->holding_zero = ~w->holding_one & 1;
}
if ((flags & HYBRID_FLAG) && ((flags & MONO_DATA) || !(csamples & 1)))
update_error_limit (w, flags);
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 (!c->error_limit)
mid = read_code (bs, high - low) + low;
else while (high - low > c->error_limit) {
if (getbit (bs))
mid = (high + (low = mid) + 1) >> 1;
else
mid = ((high = mid - 1) + low + 1) >> 1;
}
*buffer++ = getbit (bs) ? ~mid : mid;
if (flags & HYBRID_BITRATE)
c->slow_level = c->slow_level - ((c->slow_level + SLO) >> SLS) + mylog2 (mid);
}
return (flags & MONO_DATA) ? csamples : (csamples / 2);
}
// 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 uint32_t read_code (Bitstream *bs, uint32_t maxcode)
{
int bitcount = count_bits (maxcode);
uint32_t 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 mylog2 (uint32_t 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 log2 for the specified 32-bit signed value.
// All input values are valid and the return values are in the range of
// +/- 8192.
int log2s (int32_t value)
{
return (value < 0) ? -mylog2 (-value) : mylog2 (value);
}
// 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).
int32_t exp2s (int log)
{
uint32_t 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 (signed char weight)
{
int result;
if ((result = (int) weight << 3) > 0)
result += (result + 64) >> 7;
return result;
}

@ -0,0 +1,350 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 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 uint32_t sample_rates [] = { 6000, 8000, 9600, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000 };
///////////////////////////// executable code ////////////////////////////////
static uint32_t read_next_header (read_stream infile, void *user_data, 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).
int WavpackOpenFileInput (WavpackContext *wpc, read_stream infile, void *user_data, char *error)
{
WavpackStream *wps = &wpc->stream;
uint32_t bcount;
//CLEAR (wpc);
wpc->infile = infile;
wpc->user_data = user_data;
wpc->total_samples = (uint32_t) -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, wpc->user_data, &wps->wphdr);
if (bcount == (uint32_t) -1) {
strcpy (error, "not compatible with this version of WavPack file!");
return 0;
}
if (wps->wphdr.block_samples && wps->wphdr.total_samples != (uint32_t) -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 0;
}
}
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.flags & FLOAT_DATA) {
wpc->config.bytes_per_sample = 3;
wpc->config.bits_per_sample = 24;
}
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 1;
}
// 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 (but will provided
// in 24-bit integers for convenience)
// 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 will be returned as 24-bit integers (and may
// also be clipped). 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.
uint32_t WavpackUnpackSamples (WavpackContext *wpc, int32_t *buffer, uint32_t samples)
{
WavpackStream *wps = &wpc->stream;
uint32_t 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, wpc->user_data, &wps->wphdr);
if (bcount == (uint32_t) -1)
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
uint32_t WavpackGetNumSamples (WavpackContext *wpc)
{
return wpc ? wpc->total_samples : (uint32_t) -1;
}
// Get the current sample index position, or -1 if unknown
uint32_t WavpackGetSampleIndex (WavpackContext *wpc)
{
if (wpc)
return wpc->stream.sample_index;
return (uint32_t) -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
uint32_t 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 uint32_t read_next_header (read_stream infile, void *user_data, WavpackHeader *wphdr)
{
char buffer [sizeof (*wphdr)], *sp = buffer + sizeof (*wphdr), *ep = sp;
uint32_t bytes_skipped = 0;
int bleft;
while (1) {
if (sp < ep) {
bleft = ep - sp;
memcpy (buffer, sp, bleft);
}
else
bleft = 0;
if (infile (user_data, buffer + bleft, sizeof (*wphdr) - bleft) != (int32_t) 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] >= (MIN_STREAM_VERS & 0xff) && sp [4] <= (MAX_STREAM_VERS & 0xff)) {
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) > 1048576L)
return -1;
}
}

@ -0,0 +1,200 @@
////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2006 Conifer Software. //
// All Rights Reserved. //
// Distributed under the BSD Software License (see license.txt) //
////////////////////////////////////////////////////////////////////////////
// wv_filter.c
// This is the main module for the demonstration WavPack command-line
// decoder filter. It uses the tiny "hardware" version of the decoder and
// accepts WavPack files on stdin and outputs a standard MS wav file to
// stdout. Note that this involves converting the data to little-endian
// (if the executing processor is not), possibly packing the data into
// fewer bytes per sample, and generating an appropriate riff wav header.
// Note that this is NOT the copy of the RIFF header that might be stored
// in the file, and any additional RIFF information and tags are lost.
// See wputils.c for further limitations.
#include "wavpack.h"
#if defined(WIN32)
#include <io.h>
#include <fcntl.h>
#endif
#include <string.h>
// These structures are used to place a wav riff header at the beginning of
// the output.
typedef struct {
char ckID [4];
uint32_t ckSize;
char formType [4];
} RiffChunkHeader;
typedef struct {
char ckID [4];
uint32_t ckSize;
} ChunkHeader;
#define ChunkHeaderFormat "4L"
typedef struct {
ushort FormatTag, NumChannels;
uint32_t SampleRate, BytesPerSecond;
ushort BlockAlign, BitsPerSample;
} WaveHeader;
#define WaveHeaderFormat "SSLLSS"
static uchar *format_samples (int bps, uchar *dst, int32_t *src, uint32_t samcnt);
static int32_t read_bytes (void *buff, int32_t bcount);
static int32_t temp_buffer [256];
int main ()
{
ChunkHeader FormatChunkHeader, DataChunkHeader;
RiffChunkHeader RiffChunkHeader;
WaveHeader WaveHeader;
uint32_t total_unpacked_samples = 0, total_samples;
int num_channels, bps;
WavpackContext *wpc;
char error [80];
#if defined(WIN32)
setmode (fileno (stdin), O_BINARY);
setmode (fileno (stdout), O_BINARY);
#endif
wpc = WavpackOpenFileInput (read_bytes, error);
if (!wpc) {
fputs (error, stderr);
fputs ("\n", stderr);
return 1;
}
num_channels = WavpackGetReducedChannels (wpc);
total_samples = WavpackGetNumSamples (wpc);
bps = WavpackGetBytesPerSample (wpc);
strncpy (RiffChunkHeader.ckID, "RIFF", sizeof (RiffChunkHeader.ckID));
RiffChunkHeader.ckSize = total_samples * num_channels * bps + sizeof (ChunkHeader) * 2 + sizeof (WaveHeader) + 4;
strncpy (RiffChunkHeader.formType, "WAVE", sizeof (RiffChunkHeader.formType));
strncpy (FormatChunkHeader.ckID, "fmt ", sizeof (FormatChunkHeader.ckID));
FormatChunkHeader.ckSize = sizeof (WaveHeader);
WaveHeader.FormatTag = 1;
WaveHeader.NumChannels = num_channels;
WaveHeader.SampleRate = WavpackGetSampleRate (wpc);
WaveHeader.BlockAlign = num_channels * bps;
WaveHeader.BytesPerSecond = WaveHeader.SampleRate * WaveHeader.BlockAlign;
WaveHeader.BitsPerSample = WavpackGetBitsPerSample (wpc);
strncpy (DataChunkHeader.ckID, "data", sizeof (DataChunkHeader.ckID));
DataChunkHeader.ckSize = total_samples * num_channels * bps;
native_to_little_endian (&RiffChunkHeader, ChunkHeaderFormat);
native_to_little_endian (&FormatChunkHeader, ChunkHeaderFormat);
native_to_little_endian (&WaveHeader, WaveHeaderFormat);
native_to_little_endian (&DataChunkHeader, ChunkHeaderFormat);
if (!fwrite (&RiffChunkHeader, sizeof (RiffChunkHeader), 1, stdout) ||
!fwrite (&FormatChunkHeader, sizeof (FormatChunkHeader), 1, stdout) ||
!fwrite (&WaveHeader, sizeof (WaveHeader), 1, stdout) ||
!fwrite (&DataChunkHeader, sizeof (DataChunkHeader), 1, stdout)) {
fputs ("can't write .WAV data, disk probably full!\n", stderr);
return 1;
}
while (1) {
uint32_t samples_unpacked;
samples_unpacked = WavpackUnpackSamples (wpc, temp_buffer, 256 / num_channels);
total_unpacked_samples += samples_unpacked;
if (samples_unpacked) {
format_samples (bps, (uchar *) temp_buffer, temp_buffer, samples_unpacked *= num_channels);
if (fwrite (temp_buffer, bps, samples_unpacked, stdout) != samples_unpacked) {
fputs ("can't write .WAV data, disk probably full!\n", stderr);
return 1;
}
}
if (!samples_unpacked)
break;
}
fflush (stdout);
if (WavpackGetNumSamples (wpc) != (uint32_t) -1 &&
total_unpacked_samples != WavpackGetNumSamples (wpc)) {
fputs ("incorrect number of samples!\n", stderr);
return 1;
}
if (WavpackGetNumErrors (wpc)) {
fputs ("crc errors detected!\n", stderr);
return 1;
}
return 0;
}
static int32_t read_bytes (void *buff, int32_t bcount)
{
return fread (buff, 1, bcount, stdin);
}
// Reformat samples from longs in processor's native endian mode to
// little-endian data with (possibly) less than 4 bytes / sample.
static uchar *format_samples (int bps, uchar *dst, int32_t *src, uint32_t samcnt)
{
int32_t temp;
switch (bps) {
case 1:
while (samcnt--)
*dst++ = *src++ + 128;
break;
case 2:
while (samcnt--) {
*dst++ = (uchar)(temp = *src++);
*dst++ = (uchar)(temp >> 8);
}
break;
case 3:
while (samcnt--) {
*dst++ = (uchar)(temp = *src++);
*dst++ = (uchar)(temp >> 8);
*dst++ = (uchar)(temp >> 16);
}
break;
case 4:
while (samcnt--) {
*dst++ = (uchar)(temp = *src++);
*dst++ = (uchar)(temp >> 8);
*dst++ = (uchar)(temp >> 16);
*dst++ = (uchar)(temp >> 24);
}
break;
}
return dst;
}

@ -4,9 +4,9 @@
idf_component_register( idf_component_register(
SRCS "dr_flac.cpp" "codec.cpp" "mad.cpp" "opus.cpp" "vorbis.cpp" SRCS "dr_flac.cpp" "codec.cpp" "mad.cpp" "opus.cpp" "vorbis.cpp"
"source_buffer.cpp" "sample.cpp" "wav.cpp" "native.cpp" "source_buffer.cpp" "sample.cpp" "wav.cpp" "native.cpp" "wavpack.cpp"
INCLUDE_DIRS "include" INCLUDE_DIRS "include"
REQUIRES "result" "libmad" "drflac" "tremor" "opusfile" "memory" "util" REQUIRES "result" "libmad" "drflac" "tremor" "opusfile" "memory" "util"
"komihash") "komihash" "wavpack")
target_compile_options("${COMPONENT_LIB}" PRIVATE ${EXTRA_WARNINGS}) target_compile_options("${COMPONENT_LIB}" PRIVATE ${EXTRA_WARNINGS})

@ -16,6 +16,7 @@
#include "types.hpp" #include "types.hpp"
#include "vorbis.hpp" #include "vorbis.hpp"
#include "wav.hpp" #include "wav.hpp"
#include "wavpack.hpp"
namespace codecs { namespace codecs {
@ -33,6 +34,8 @@ auto StreamTypeToString(StreamType t) -> std::string {
return "Opus"; return "Opus";
case StreamType::kNative: case StreamType::kNative:
return "Native"; return "Native";
case StreamType::kWavPack:
return "WavPack";
default: default:
return ""; return "";
} }
@ -52,6 +55,8 @@ auto CreateCodecForType(StreamType type) -> std::optional<ICodec*> {
return new WavDecoder(); return new WavDecoder();
case StreamType::kNative: case StreamType::kNative:
return new NativeDecoder(); return new NativeDecoder();
case StreamType::kWavPack:
return new WavPackDecoder();
default: default:
return {}; return {};
} }

@ -17,6 +17,7 @@ enum class StreamType {
kOpus, kOpus,
kWav, kWav,
kNative, kNative,
kWavPack,
}; };
auto StreamTypeToString(StreamType t) -> std::string; auto StreamTypeToString(StreamType t) -> std::string;

@ -0,0 +1,46 @@
/*
* Copyright 2025 ayumi <ayumi@noreply.codeberg.org>
*
* SPDX-License-Identifier: GPL-3.0-only
*/
#pragma once
#include <cstddef>
#include <cstdint>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include "wavpack.h"
#include "sample.hpp"
#include "codec.hpp"
namespace codecs {
class WavPackDecoder : public ICodec {
public:
WavPackDecoder();
~WavPackDecoder();
auto OpenStream(std::shared_ptr<IStream> input, uint32_t offset)
-> cpp::result<OutputFormat, Error> override;
auto DecodeTo(std::span<sample::Sample> destination)
-> cpp::result<OutputInfo, Error> override;
WavPackDecoder(const WavPackDecoder&) = delete;
WavPackDecoder& operator=(const WavPackDecoder&) = delete;
private:
std::shared_ptr<IStream> input_;
WavpackContext wavpack_;
int32_t *buf_;
uint8_t bitdepth_;
uint8_t channels_;
size_t size_;
};
} // namespace codecs

@ -0,0 +1,161 @@
/*
* Copyright 2025 ayumi <ayumi@noreply.codeberg.org>
*
* SPDX-License-Identifier: GPL-3.0-only
*/
#include "wavpack.hpp"
#include <cstdint>
#include <cstring>
#include <algorithm>
#include <optional>
#include "esp_heap_caps.h"
#include "codec.hpp"
#include "esp_log.h"
#include "result.hpp"
#include "sample.hpp"
#include "types.hpp"
namespace codecs {
[[maybe_unused]] static constexpr const char kTag[] = "wavpack";
// kBufSize and audio::kCodecBufferLength must be equal
static constexpr const size_t kBufSize = 2048;
static inline constexpr auto loadLe16(std::byte* data) -> uint16_t {
return *reinterpret_cast<uint16_t*>(data);
}
static inline constexpr auto loadLe32(std::byte* data) -> uint32_t {
return *reinterpret_cast<uint32_t*>(data);
}
static auto readProc(void* data, void* buf, long size) -> long {
IStream* stream = static_cast<IStream*>(data);
const int32_t res = stream->Read({
static_cast<std::byte*>(buf),
static_cast<std::span<std::byte>::size_type>(size)
});
return res < 0 ? 0 : res;
}
WavPackDecoder::WavPackDecoder() : input_(), buf_() {
buf_ = static_cast<int32_t*>(
heap_caps_malloc(
kBufSize * sizeof(int32_t),
MALLOC_CAP_INTERNAL | MALLOC_CAP_CACHE_ALIGNED
));
}
WavPackDecoder::~WavPackDecoder() {
heap_caps_free(buf_);
}
auto WavPackDecoder::OpenStream(std::shared_ptr<IStream> input, uint32_t offset)
-> cpp::result<OutputFormat, ICodec::Error> {
char error[80];
input_ = input;
wavpack_ = {};
if (!WavpackOpenFileInput(&wavpack_, readProc, input_.get(), error)) {
ESP_LOGE(kTag, "WavpackOpenFileInput: %s", error);
return cpp::fail(Error::kMalformedData);
}
channels_ = WavpackGetReducedChannels(&wavpack_);
bitdepth_ = WavpackGetBitsPerSample(&wavpack_);
size_ = kBufSize / channels_;
const std::optional total = WavpackGetNumSamples(&wavpack_) == -1
? std::nullopt
: std::optional(
static_cast<uint64_t>(WavpackGetNumSamples(&wavpack_)) * channels_
);
const auto rate = WavpackGetSampleRate(&wavpack_);
if (offset && total && input_.get()->CanSeek()) {
const uint32_t want = offset * rate - 1;
if (total < want) {
ESP_LOGE(kTag, "seeking: offset points beyond the end of the file");
return cpp::fail(Error::kInternalError);
}
uint32_t target;
input_->SeekTo(0, IStream::SeekFrom::kStartOfStream);
while (true) {
std::byte header[32];
input_->Read(header);
if (memcmp(header, "wvpk", 4) != 0) {
ESP_LOGE(kTag, "seeking: header expected, but not found");
return cpp::fail(Error::kMalformedData);
}
const uint32_t size = loadLe32(header + 4);
const uint16_t version = loadLe16(header + 8);
if (version < 0x402 || version > 0x410) {
ESP_LOGE(kTag, "seeking: bad WavPack version (%x)", version);
return cpp::fail(Error::kMalformedData);
}
const uint32_t blockIndex = loadLe32(header + 16);
const uint32_t blockSamples = loadLe32(header + 20);
if (want >= blockIndex && want <= blockIndex + blockSamples) {
input_->SeekTo(-32, IStream::SeekFrom::kCurrentPosition);
target = want - blockIndex;
break;
}
input_->SeekTo(size - 24, IStream::SeekFrom::kCurrentPosition);
}
wavpack_ = {};
if (!WavpackOpenFileInput(&wavpack_, readProc, input_.get(), error)) {
ESP_LOGE(kTag, "WavpackOpenFileInput: %s", error);
return cpp::fail(Error::kMalformedData);
}
uint32_t samples = 0;
for (size_t i = 0, n = target / size_; i < n; i++)
samples += WavpackUnpackSamples(&wavpack_, buf_, size_);
samples += WavpackUnpackSamples(&wavpack_, buf_, target % size_);
if (WavpackGetNumErrors(&wavpack_) != 0) {
ESP_LOGE(kTag, "CRC error");
return cpp::fail(Error::kMalformedData);
} else if (samples != target || WavpackGetSampleIndex(&wavpack_) != want) {
ESP_LOGE(kTag, "seeking: seeking unsuccessful: want %lu, got %lu",
target, samples
);
return cpp::fail(Error::kInternalError);
}
} else if (offset && (!total || !input_.get()->CanSeek())) {
ESP_LOGE(kTag, "seeking: can’t seek");
return cpp::fail(Error::kInternalError);
}
const auto size = input->Size();
return OutputFormat{
.num_channels = channels_,
.sample_rate_hz = rate,
.total_samples = total,
.bitrate_kbps = size && total
? std::optional(
((double)size.value() * 8.0)
/ ((double)total.value() / channels_ / rate) / 1000
)
: std::nullopt,
};
}
auto WavPackDecoder::DecodeTo(std::span<sample::Sample> output)
-> cpp::result<OutputInfo, Error> {
const auto size = std::min(size_, output.size() / channels_);
const auto samples = WavpackUnpackSamples(&wavpack_, buf_, size) * channels_;
if (WavpackGetNumErrors(&wavpack_) != 0) {
ESP_LOGE(kTag, "CRC error");
return cpp::fail(Error::kMalformedData);
}
for (size_t i = 0; i < samples; i++)
output[i] = sample::FromSigned(buf_[i], bitdepth_);
return OutputInfo{
.samples_written = samples,
.is_stream_finished = samples == 0,
};
}
} // namespace codecs

@ -88,6 +88,8 @@ auto FatfsStreamFactory::ContainerToStreamType(database::Container enc)
return codecs::StreamType::kFlac; return codecs::StreamType::kFlac;
case database::Container::kOpus: case database::Container::kOpus:
return codecs::StreamType::kOpus; return codecs::StreamType::kOpus;
case database::Container::kWavPack:
return codecs::StreamType::kWavPack;
case database::Container::kUnsupported: case database::Container::kUnsupported:
default: default:
return {}; return {};

@ -413,6 +413,9 @@ auto GenericTagParser::ReadAndParseTags(std::string_view p)
case Fopus: case Fopus:
out->encoding(Container::kOpus); out->encoding(Container::kOpus);
break; break;
case Fwavpack:
out->encoding(Container::kWavPack);
break;
default: default:
out->encoding(Container::kUnsupported); out->encoding(Container::kUnsupported);
} }

@ -63,7 +63,8 @@ class GenericTagParser : public ITagParser {
// supported audio formats here: // supported audio formats here:
// https://cooltech.zone/tangara/docs/music-library/ // https://cooltech.zone/tangara/docs/music-library/
static constexpr std::string supported_exts[] = {"flac", "mp3", "ogg", static constexpr std::string supported_exts[] = {"flac", "mp3", "ogg",
"ogx", "opus", "wav"}; "ogx", "opus", "wav",
"wv"};
}; };
} // namespace database } // namespace database

@ -45,6 +45,7 @@ enum class Container {
kOgg = 3, kOgg = 3,
kFlac = 4, kFlac = 4,
kOpus = 5, kOpus = 5,
kWavPack = 6,
}; };
enum class MediaType { enum class MediaType {

@ -37,6 +37,7 @@ list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/result")
list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/speexdsp") list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/speexdsp")
list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/tinyfsm") list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/tinyfsm")
list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/tremor") list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/tremor")
list(APPEND EXTRA_COMPONENT_DIRS "$ENV{PROJ_PATH}/lib/wavpack")
include($ENV{IDF_PATH}/tools/cmake/project.cmake) include($ENV{IDF_PATH}/tools/cmake/project.cmake)

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