cog/Frameworks/WavPack/Files/unpack.c

1418 lines
53 KiB
C

////////////////////////////////////////////////////////////////////////////
// **** 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? modules. For
// maximum efficiency, the conversion is isolated to tight loops that handle
// an entire buffer.
#include "wavpack_local.h"
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
// This flag provides faster decoding speed at the expense of more code. The
// improvement applies to 16-bit stereo lossless only.
#define FAST_DECODE
#define LOSSY_MUTE
#ifdef DEBUG_ALLOC
#define malloc malloc_db
#define realloc realloc_db
#define free free_db
void *malloc_db (uint32_t size);
void *realloc_db (void *ptr, uint32_t size);
void free_db (void *ptr);
int32_t dump_alloc (void);
#endif
///////////////////////////// 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) and that the entire block has been read at
// wps->blockbuff. If a correction file is available (wpc->wvc_flag = TRUE)
// then the corresponding correction block must be read into wps->block2buff
// and its WavpackHeader has overwritten the header at wps->wphdr. This is
// where all the metadata blocks are scanned including those that contain
// bitstream data.
int unpack_init (WavpackContext *wpc)
{
WavpackStream *wps = wpc->streams [wpc->current_stream];
unsigned char *blockptr, *block2ptr;
WavpackMetadata wpmd;
wps->mute_error = FALSE;
wps->crc = wps->crc_x = 0xffffffff;
CLEAR (wps->wvbits);
CLEAR (wps->wvcbits);
CLEAR (wps->wvxbits);
CLEAR (wps->decorr_passes);
CLEAR (wps->dc);
CLEAR (wps->w);
if (!(wps->wphdr.flags & MONO_FLAG) && wpc->config.num_channels && wps->wphdr.block_samples &&
(wpc->reduced_channels == 1 || wpc->config.num_channels == 1)) {
wps->mute_error = TRUE;
return FALSE;
}
if ((wps->wphdr.flags & UNKNOWN_FLAGS) || (wps->wphdr.flags & MONO_DATA) == MONO_DATA) {
wps->mute_error = TRUE;
return FALSE;
}
blockptr = wps->blockbuff + sizeof (WavpackHeader);
while (read_metadata_buff (&wpmd, wps->blockbuff, &blockptr))
if (!process_metadata (wpc, &wpmd)) {
wps->mute_error = TRUE;
return FALSE;
}
if (wps->wphdr.block_samples && wpc->wvc_flag && wps->block2buff) {
block2ptr = wps->block2buff + sizeof (WavpackHeader);
while (read_metadata_buff (&wpmd, wps->block2buff, &block2ptr))
if (!process_metadata (wpc, &wpmd)) {
wps->mute_error = TRUE;
return FALSE;
}
}
if (wps->wphdr.block_samples && !bs_is_open (&wps->wvbits)) {
if (bs_is_open (&wps->wvcbits))
strcpy (wpc->error_message, "can't unpack correction files alone!");
wps->mute_error = TRUE;
return FALSE;
}
if (wps->wphdr.block_samples && !bs_is_open (&wps->wvxbits)) {
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;
}
if (wps->wphdr.block_samples)
wps->sample_index = wps->wphdr.block_index;
return TRUE;
}
// This function initialzes the main bitstream for audio samples, which must
// be in the "wv" file.
int init_wv_bitstream (WavpackStream *wps, WavpackMetadata *wpmd)
{
if (!wpmd->byte_length)
return FALSE;
bs_open_read (&wps->wvbits, wpmd->data, (unsigned char *) wpmd->data + wpmd->byte_length);
return TRUE;
}
// This function initialzes the "correction" bitstream for audio samples,
// which currently must be in the "wvc" file.
int init_wvc_bitstream (WavpackStream *wps, WavpackMetadata *wpmd)
{
if (!wpmd->byte_length)
return FALSE;
bs_open_read (&wps->wvcbits, wpmd->data, (unsigned char *) wpmd->data + wpmd->byte_length);
return TRUE;
}
// This function initialzes the "extra" bitstream for audio samples which
// contains the information required to losslessly decompress 32-bit float data
// or integer data that exceeds 24 bits. This bitstream is in the "wv" file
// for pure lossless data or the "wvc" file for hybrid lossless. This data
// would not be used for hybrid lossy mode. There is also a 32-bit CRC stored
// in the first 4 bytes of these blocks.
int init_wvx_bitstream (WavpackStream *wps, WavpackMetadata *wpmd)
{
unsigned char *cp = wpmd->data;
if (wpmd->byte_length <= 4)
return FALSE;
wps->crc_wvx = *cp++;
wps->crc_wvx |= (int32_t) *cp++ << 8;
wps->crc_wvx |= (int32_t) *cp++ << 16;
wps->crc_wvx |= (int32_t) *cp++ << 24;
bs_open_read (&wps->wvxbits, cp, (unsigned char *) wpmd->data + wpmd->byte_length);
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;
unsigned char *byteptr = wpmd->data;
struct decorr_pass *dpp;
if (termcnt > MAX_NTERMS)
return FALSE;
wps->num_terms = termcnt;
for (dpp = wps->decorr_passes + termcnt - 1; termcnt--; dpp--) {
dpp->term = (int)(*byteptr & 0x1f) - 5;
dpp->delta = (*byteptr++ >> 5) & 0x7;
if (!dpp->term || dpp->term < -3 || (dpp->term > MAX_TERM && dpp->term < 17) || dpp->term > 18)
return FALSE;
}
return TRUE;
}
// Read decorrelation weights from specified metadata block into the
// decorr_passes array. The weights range +/-1024, but are rounded and
// truncated to fit in signed chars for metadata storage. Weights are
// separate for the two channels and are specified from the "last" term
// (first during encode). Unspecified weights are set to zero.
int read_decorr_weights (WavpackStream *wps, WavpackMetadata *wpmd)
{
int termcnt = wpmd->byte_length, tcount;
char *byteptr = wpmd->data;
struct decorr_pass *dpp;
if (!(wps->wphdr.flags & MONO_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)
{
unsigned char *byteptr = wpmd->data;
unsigned char *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)) {
if (byteptr + (wps->wphdr.flags & MONO_DATA ? 2 : 4) > endptr)
return FALSE;
wps->dc.error [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
byteptr += 2;
if (!(wps->wphdr.flags & MONO_DATA)) {
wps->dc.error [1] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
byteptr += 2;
}
}
while (dpp-- > wps->decorr_passes && byteptr < endptr)
if (dpp->term > MAX_TERM) {
if (byteptr + (wps->wphdr.flags & MONO_DATA ? 4 : 8) > endptr)
return FALSE;
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) {
if (byteptr + 4 > endptr)
return FALSE;
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--) {
if (byteptr + (wps->wphdr.flags & MONO_DATA ? 2 : 4) > endptr)
return FALSE;
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 shaping weights from specified metadata block into the
// WavpackStream structure. Note that there must be two values (even
// for mono streams) and that the values are stored in the same
// manner as decorrelation weights. These would normally be read from
// the "correction" file and are used for lossless reconstruction of
// hybrid data.
int read_shaping_info (WavpackStream *wps, WavpackMetadata *wpmd)
{
if (wpmd->byte_length == 2) {
char *byteptr = wpmd->data;
wps->dc.shaping_acc [0] = (int32_t) restore_weight (*byteptr++) << 16;
wps->dc.shaping_acc [1] = (int32_t) restore_weight (*byteptr++) << 16;
return TRUE;
}
else if (wpmd->byte_length >= (wps->wphdr.flags & MONO_DATA ? 4 : 8)) {
unsigned char *byteptr = wpmd->data;
wps->dc.error [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
wps->dc.shaping_acc [0] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
byteptr += 4;
if (!(wps->wphdr.flags & MONO_DATA)) {
wps->dc.error [1] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
wps->dc.shaping_acc [1] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
byteptr += 4;
}
if (wpmd->byte_length == (wps->wphdr.flags & MONO_DATA ? 6 : 12)) {
wps->dc.shaping_delta [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
if (!(wps->wphdr.flags & MONO_DATA))
wps->dc.shaping_delta [1] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
}
return TRUE;
}
return FALSE;
}
// 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;
unsigned char *byteptr = wpmd->data;
uint32_t mask = 0;
if (!bytecnt || bytecnt > 6)
return FALSE;
if (!wpc->config.num_channels) {
if (bytecnt == 6) {
wpc->config.num_channels = (byteptr [0] | ((byteptr [2] & 0xf) << 8)) + 1;
wpc->max_streams = (byteptr [1] | ((byteptr [2] & 0xf0) << 4)) + 1;
if (wpc->config.num_channels < wpc->max_streams)
return FALSE;
byteptr += 3;
mask = *byteptr++;
mask |= (uint32_t) *byteptr++ << 8;
mask |= (uint32_t) *byteptr << 16;
}
else {
wpc->config.num_channels = *byteptr++;
while (--bytecnt) {
mask |= (uint32_t) *byteptr++ << shift;
shift += 8;
}
}
if (wpc->config.num_channels > wpc->max_streams * 2)
return FALSE;
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;
unsigned char *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;
if (bytecnt >= 4 && (wpc->config.flags & CONFIG_EXTRA_MODE))
wpc->config.xmode = *byteptr;
}
return TRUE;
}
// Read non-standard sampling rate from metadata.
int read_sample_rate (WavpackContext *wpc, WavpackMetadata *wpmd)
{
int bytecnt = wpmd->byte_length;
unsigned char *byteptr = wpmd->data;
if (bytecnt == 3) {
wpc->config.sample_rate = (int32_t) *byteptr++;
wpc->config.sample_rate |= (int32_t) *byteptr++ << 8;
wpc->config.sample_rate |= (int32_t) *byteptr++ << 16;
}
return TRUE;
}
// Read wrapper data from metadata. Currently, this consists of the RIFF
// header and trailer that wav files contain around the audio data but could
// be used for other formats as well. Because WavPack files contain all the
// information required for decoding and playback, this data can probably
// be ignored except when an exact wavefile restoration is needed.
int read_wrapper_data (WavpackContext *wpc, WavpackMetadata *wpmd)
{
if ((wpc->open_flags & OPEN_WRAPPER) && wpc->wrapper_bytes < MAX_WRAPPER_BYTES) {
wpc->wrapper_data = realloc (wpc->wrapper_data, wpc->wrapper_bytes + wpmd->byte_length);
memcpy (wpc->wrapper_data + wpc->wrapper_bytes, wpmd->data, wpmd->byte_length);
wpc->wrapper_bytes += wpmd->byte_length;
}
return TRUE;
}
#ifndef NO_UNPACK
// This monster actually unpacks the WavPack bitstream(s) into the specified
// buffer as 32-bit integers or floats (depending on orignal data). Lossy
// samples will be clipped to their original limits (i.e. 8-bit samples are
// clipped to -128/+127) but are still returned in longs. It is up to the
// caller to potentially reformat this for the final output including any
// multichannel distribution, block alignment or endian compensation. The
// function unpack_init() must have been called and the entire WavPack block
// must still be visible (although wps->blockbuff will not be accessed again).
// For maximum clarity, the function is broken up into segments that handle
// various modes. This makes for a few extra infrequent flag checks, but
// makes the code easier to follow because the nesting does not become so
// deep. For maximum efficiency, the conversion is isolated to tight loops
// that handle an entire buffer. The function returns the total number of
// samples unpacked, which can be less than the number requested if an error
// occurs or the end of the block is reached.
static void decorr_stereo_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void decorr_stereo_pass_i (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void decorr_stereo_pass_1717 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void decorr_stereo_pass_1718 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void decorr_stereo_pass_1818 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void decorr_stereo_pass_nn (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
static void fixup_samples (WavpackContext *wpc, int32_t *buffer, uint32_t sample_count);
int32_t unpack_samples (WavpackContext *wpc, int32_t *buffer, uint32_t sample_count)
{
WavpackStream *wps = wpc->streams [wpc->current_stream];
uint32_t flags = wps->wphdr.flags, crc = wps->crc, i;
int32_t mute_limit = (int32_t)((1L << ((flags & MAG_MASK) >> MAG_LSB)) + 2);
int32_t correction [2], read_word, *bptr;
struct decorr_pass *dpp;
int tcount, m = 0;
if (wps->sample_index + sample_count > wps->wphdr.block_index + wps->wphdr.block_samples)
sample_count = wps->wphdr.block_index + wps->wphdr.block_samples - wps->sample_index;
if (wps->mute_error) {
if (wpc->reduced_channels == 1 || wpc->config.num_channels == 1 || (flags & MONO_FLAG))
memset (buffer, 0, sample_count * 4);
else
memset (buffer, 0, sample_count * 8);
wps->sample_index += sample_count;
return sample_count;
}
if ((flags & HYBRID_FLAG) && !wps->block2buff)
mute_limit *= 2;
//////////////// handle lossless or hybrid lossy mono data /////////////////
if (!wps->block2buff && (flags & MONO_DATA)) {
int32_t *eptr = buffer + sample_count;
if (flags & HYBRID_FLAG) {
i = sample_count;
for (bptr = buffer; bptr < eptr;)
if ((*bptr++ = get_word (wps, 0, NULL)) == WORD_EOF) {
i = (uint32_t)(bptr - buffer);
break;
}
}
else
i = get_words_lossless (wps, buffer, sample_count);
for (bptr = buffer; bptr < eptr;) {
read_word = *bptr;
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
int32_t sam, temp;
int k;
if (dpp->term > MAX_TERM) {
if (dpp->term & 1)
sam = 2 * dpp->samples_A [0] - dpp->samples_A [1];
else
sam = dpp->samples_A [0] + ((dpp->samples_A [0] - dpp->samples_A [1]) >> 1);
dpp->samples_A [1] = dpp->samples_A [0];
k = 0;
}
else {
sam = dpp->samples_A [m];
k = (m + dpp->term) & (MAX_TERM - 1);
}
temp = apply_weight (dpp->weight_A, sam) + read_word;
update_weight (dpp->weight_A, dpp->delta, sam, read_word);
dpp->samples_A [k] = read_word = temp;
}
if (labs (read_word) > mute_limit) {
i = (uint32_t)(bptr - buffer);
break;
}
m = (m + 1) & (MAX_TERM - 1);
crc += (crc << 1) + (*bptr++ = read_word);
}
}
/////////////// handle lossless or hybrid lossy stereo data ///////////////
else if (!wps->block2buff && !(flags & MONO_DATA)) {
int32_t *eptr = buffer + (sample_count * 2);
if (flags & HYBRID_FLAG) {
i = sample_count;
for (bptr = buffer; bptr < eptr; bptr += 2)
if ((bptr [0] = get_word (wps, 0, NULL)) == WORD_EOF ||
(bptr [1] = get_word (wps, 1, NULL)) == WORD_EOF) {
i = (uint32_t)(bptr - buffer) / 2;
break;
}
}
else
i = get_words_lossless (wps, buffer, sample_count);
#ifdef FAST_DECODE
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
if (((flags & MAG_MASK) >> MAG_LSB) >= 16)
decorr_stereo_pass (dpp, buffer, sample_count);
else if (tcount && dpp [0].term == 17 && dpp [1].term == 17) {
decorr_stereo_pass_1717 (dpp, buffer, sample_count);
tcount--;
dpp++;
}
else if (tcount && dpp [0].term == 17 && dpp [1].term == 18) {
decorr_stereo_pass_1718 (dpp, buffer, sample_count);
tcount--;
dpp++;
}
else if (tcount && dpp [0].term == 18 && dpp [1].term == 18) {
decorr_stereo_pass_1818 (dpp, buffer, sample_count);
tcount--;
dpp++;
}
else if (tcount && dpp [0].term >= 1 && dpp [0].term <= 7 &&
dpp [1].term >= 1 && dpp [1].term <= 7) {
decorr_stereo_pass_nn (dpp, buffer, sample_count);
tcount--;
dpp++;
}
else
decorr_stereo_pass_i (dpp, buffer, sample_count);
#else
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
decorr_stereo_pass (dpp, buffer, sample_count);
#endif
if (flags & JOINT_STEREO)
for (bptr = buffer; bptr < eptr; bptr += 2) {
bptr [0] += (bptr [1] -= (bptr [0] >> 1));
crc += (crc << 3) + (bptr [0] << 1) + bptr [0] + bptr [1];
}
else
for (bptr = buffer; bptr < eptr; bptr += 2)
crc += (crc << 3) + (bptr [0] << 1) + bptr [0] + bptr [1];
for (bptr = buffer; bptr < eptr; bptr += 16)
if (labs (bptr [0]) > mute_limit || labs (bptr [1]) > mute_limit) {
i = (uint32_t)(bptr - buffer) / 2;
break;
}
m = sample_count & (MAX_TERM - 1);
}
/////////////////// handle hybrid lossless mono data ////////////////////
else if ((flags & HYBRID_FLAG) && (flags & MONO_DATA))
for (bptr = buffer, i = 0; i < sample_count; ++i) {
if ((read_word = get_word (wps, 0, correction)) == WORD_EOF)
break;
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
int32_t sam, temp;
int k;
if (dpp->term > MAX_TERM) {
if (dpp->term & 1)
sam = 2 * dpp->samples_A [0] - dpp->samples_A [1];
else
sam = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
dpp->samples_A [1] = dpp->samples_A [0];
k = 0;
}
else {
sam = dpp->samples_A [m];
k = (m + dpp->term) & (MAX_TERM - 1);
}
temp = apply_weight (dpp->weight_A, sam) + read_word;
update_weight (dpp->weight_A, dpp->delta, sam, read_word);
dpp->samples_A [k] = read_word = temp;
}
m = (m + 1) & (MAX_TERM - 1);
if (flags & HYBRID_SHAPE) {
int shaping_weight = (wps->dc.shaping_acc [0] += wps->dc.shaping_delta [0]) >> 16;
int32_t temp = -apply_weight (shaping_weight, wps->dc.error [0]);
if ((flags & NEW_SHAPING) && shaping_weight < 0 && temp) {
if (temp == wps->dc.error [0])
temp = (temp < 0) ? temp + 1 : temp - 1;
wps->dc.error [0] = temp - correction [0];
}
else
wps->dc.error [0] = -correction [0];
read_word += correction [0] - temp;
}
else
read_word += correction [0];
crc += (crc << 1) + read_word;
#ifdef LOSSY_MUTE
if (labs (read_word) > mute_limit)
break;
#endif
*bptr++ = read_word;
}
//////////////////// handle hybrid lossless stereo data ///////////////////
else if (wps->block2buff && !(flags & MONO_DATA))
for (bptr = buffer, i = 0; i < sample_count; ++i) {
int32_t left, right, left2, right2;
int32_t left_c = 0, right_c = 0;
if ((left = get_word (wps, 0, correction)) == WORD_EOF ||
(right = get_word (wps, 1, correction + 1)) == WORD_EOF)
break;
if (flags & CROSS_DECORR) {
left_c = left + correction [0];
right_c = right + correction [1];
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
int32_t sam_A, sam_B;
if (dpp->term > 0) {
if (dpp->term > MAX_TERM) {
if (dpp->term & 1) {
sam_A = 2 * dpp->samples_A [0] - dpp->samples_A [1];
sam_B = 2 * dpp->samples_B [0] - dpp->samples_B [1];
}
else {
sam_A = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
sam_B = (3 * dpp->samples_B [0] - dpp->samples_B [1]) >> 1;
}
}
else {
sam_A = dpp->samples_A [m];
sam_B = dpp->samples_B [m];
}
left_c += apply_weight (dpp->weight_A, sam_A);
right_c += apply_weight (dpp->weight_B, sam_B);
}
else if (dpp->term == -1) {
left_c += apply_weight (dpp->weight_A, dpp->samples_A [0]);
right_c += apply_weight (dpp->weight_B, left_c);
}
else {
right_c += apply_weight (dpp->weight_B, dpp->samples_B [0]);
if (dpp->term == -3)
left_c += apply_weight (dpp->weight_A, dpp->samples_A [0]);
else
left_c += apply_weight (dpp->weight_A, right_c);
}
}
if (flags & JOINT_STEREO)
left_c += (right_c -= (left_c >> 1));
}
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
int32_t sam_A, sam_B;
if (dpp->term > 0) {
int k;
if (dpp->term > MAX_TERM) {
if (dpp->term & 1) {
sam_A = 2 * dpp->samples_A [0] - dpp->samples_A [1];
sam_B = 2 * dpp->samples_B [0] - dpp->samples_B [1];
}
else {
sam_A = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
sam_B = (3 * dpp->samples_B [0] - dpp->samples_B [1]) >> 1;
}
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_B [1] = dpp->samples_B [0];
k = 0;
}
else {
sam_A = dpp->samples_A [m];
sam_B = dpp->samples_B [m];
k = (m + dpp->term) & (MAX_TERM - 1);
}
left2 = apply_weight (dpp->weight_A, sam_A) + left;
right2 = apply_weight (dpp->weight_B, sam_B) + right;
update_weight (dpp->weight_A, dpp->delta, sam_A, left);
update_weight (dpp->weight_B, dpp->delta, sam_B, right);
dpp->samples_A [k] = left = left2;
dpp->samples_B [k] = right = right2;
}
else if (dpp->term == -1) {
left2 = left + apply_weight (dpp->weight_A, dpp->samples_A [0]);
update_weight_clip (dpp->weight_A, dpp->delta, dpp->samples_A [0], left);
left = left2;
right2 = right + apply_weight (dpp->weight_B, left2);
update_weight_clip (dpp->weight_B, dpp->delta, left2, right);
dpp->samples_A [0] = right = right2;
}
else {
right2 = right + apply_weight (dpp->weight_B, dpp->samples_B [0]);
update_weight_clip (dpp->weight_B, dpp->delta, dpp->samples_B [0], right);
right = right2;
if (dpp->term == -3) {
right2 = dpp->samples_A [0];
dpp->samples_A [0] = right;
}
left2 = left + apply_weight (dpp->weight_A, right2);
update_weight_clip (dpp->weight_A, dpp->delta, right2, left);
dpp->samples_B [0] = left = left2;
}
}
m = (m + 1) & (MAX_TERM - 1);
if (!(flags & CROSS_DECORR)) {
left_c = left + correction [0];
right_c = right + correction [1];
if (flags & JOINT_STEREO)
left_c += (right_c -= (left_c >> 1));
}
if (flags & JOINT_STEREO)
left += (right -= (left >> 1));
if (flags & HYBRID_SHAPE) {
int shaping_weight;
int32_t temp;
correction [0] = left_c - left;
shaping_weight = (wps->dc.shaping_acc [0] += wps->dc.shaping_delta [0]) >> 16;
temp = -apply_weight (shaping_weight, wps->dc.error [0]);
if ((flags & NEW_SHAPING) && shaping_weight < 0 && temp) {
if (temp == wps->dc.error [0])
temp = (temp < 0) ? temp + 1 : temp - 1;
wps->dc.error [0] = temp - correction [0];
}
else
wps->dc.error [0] = -correction [0];
left = left_c - temp;
correction [1] = right_c - right;
shaping_weight = (wps->dc.shaping_acc [1] += wps->dc.shaping_delta [1]) >> 16;
temp = -apply_weight (shaping_weight, wps->dc.error [1]);
if ((flags & NEW_SHAPING) && shaping_weight < 0 && temp) {
if (temp == wps->dc.error [1])
temp = (temp < 0) ? temp + 1 : temp - 1;
wps->dc.error [1] = temp - correction [1];
}
else
wps->dc.error [1] = -correction [1];
right = right_c - temp;
}
else {
left = left_c;
right = right_c;
}
#ifdef LOSSY_MUTE
if (labs (left) > mute_limit || labs (right) > mute_limit)
break;
#endif
crc += (crc << 3) + (left << 1) + left + right;
*bptr++ = left;
*bptr++ = right;
}
else
i = 0; /* this line can't execute, but suppresses compiler warning */
if (i != sample_count) {
memset (buffer, 0, sample_count * (flags & MONO_FLAG ? 4 : 8));
wps->mute_error = TRUE;
i = sample_count;
if (bs_is_open (&wps->wvxbits))
bs_close_read (&wps->wvxbits);
}
if (m)
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
if (dpp->term > 0 && dpp->term <= MAX_TERM) {
int32_t temp_A [MAX_TERM], temp_B [MAX_TERM];
int k;
memcpy (temp_A, dpp->samples_A, sizeof (dpp->samples_A));
memcpy (temp_B, dpp->samples_B, sizeof (dpp->samples_B));
for (k = 0; k < MAX_TERM; k++) {
dpp->samples_A [k] = temp_A [m];
dpp->samples_B [k] = temp_B [m];
m = (m + 1) & (MAX_TERM - 1);
}
}
fixup_samples (wpc, buffer, i);
if ((flags & FLOAT_DATA) && (wpc->open_flags & OPEN_NORMALIZE))
WavpackFloatNormalize (buffer, (flags & MONO_DATA) ? i : i * 2,
127 - wps->float_norm_exp + wpc->norm_offset);
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;
}
// General function to perform stereo decorrelation pass on specified buffer
// (although since this is the reverse function it might technically be called
// "correlation" instead). This version handles all sample resolutions and
// weight deltas. The dpp->samples_X[] data is *not* returned normalized for
// term values 1-8, so it should be normalized if it is going to be used to
// call this function again.
static void decorr_stereo_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t *bptr, *eptr = buffer + (sample_count * 2);
int m, k;
switch (dpp->term) {
case 17:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam, tmp;
sam = 2 * dpp->samples_A [0] - dpp->samples_A [1];
dpp->samples_A [1] = dpp->samples_A [0];
bptr [0] = dpp->samples_A [0] = apply_weight (dpp->weight_A, sam) + (tmp = bptr [0]);
update_weight (dpp->weight_A, dpp->delta, sam, tmp);
sam = 2 * dpp->samples_B [0] - dpp->samples_B [1];
dpp->samples_B [1] = dpp->samples_B [0];
bptr [1] = dpp->samples_B [0] = apply_weight (dpp->weight_B, sam) + (tmp = bptr [1]);
update_weight (dpp->weight_B, dpp->delta, sam, tmp);
}
break;
case 18:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam, tmp;
sam = dpp->samples_A [0] + ((dpp->samples_A [0] - dpp->samples_A [1]) >> 1);
dpp->samples_A [1] = dpp->samples_A [0];
bptr [0] = dpp->samples_A [0] = apply_weight (dpp->weight_A, sam) + (tmp = bptr [0]);
update_weight (dpp->weight_A, dpp->delta, sam, tmp);
sam = dpp->samples_B [0] + ((dpp->samples_B [0] - dpp->samples_B [1]) >> 1);
dpp->samples_B [1] = dpp->samples_B [0];
bptr [1] = dpp->samples_B [0] = apply_weight (dpp->weight_B, sam) + (tmp = bptr [1]);
update_weight (dpp->weight_B, dpp->delta, sam, tmp);
}
break;
default:
for (m = 0, k = dpp->term & (MAX_TERM - 1), bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = dpp->samples_A [m];
dpp->samples_A [k] = apply_weight (dpp->weight_A, sam) + bptr [0];
update_weight (dpp->weight_A, dpp->delta, sam, bptr [0]);
bptr [0] = dpp->samples_A [k];
sam = dpp->samples_B [m];
dpp->samples_B [k] = apply_weight (dpp->weight_B, sam) + bptr [1];
update_weight (dpp->weight_B, dpp->delta, sam, bptr [1]);
bptr [1] = dpp->samples_B [k];
m = (m + 1) & (MAX_TERM - 1);
k = (k + 1) & (MAX_TERM - 1);
}
break;
case -1:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = bptr [0] + apply_weight (dpp->weight_A, dpp->samples_A [0]);
update_weight_clip (dpp->weight_A, dpp->delta, dpp->samples_A [0], bptr [0]);
bptr [0] = sam;
dpp->samples_A [0] = bptr [1] + apply_weight (dpp->weight_B, sam);
update_weight_clip (dpp->weight_B, dpp->delta, sam, bptr [1]);
bptr [1] = dpp->samples_A [0];
}
break;
case -2:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = bptr [1] + apply_weight (dpp->weight_B, dpp->samples_B [0]);
update_weight_clip (dpp->weight_B, dpp->delta, dpp->samples_B [0], bptr [1]);
bptr [1] = sam;
dpp->samples_B [0] = bptr [0] + apply_weight (dpp->weight_A, sam);
update_weight_clip (dpp->weight_A, dpp->delta, sam, bptr [0]);
bptr [0] = dpp->samples_B [0];
}
break;
case -3:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam_A, sam_B;
sam_A = bptr [0] + apply_weight (dpp->weight_A, dpp->samples_A [0]);
update_weight_clip (dpp->weight_A, dpp->delta, dpp->samples_A [0], bptr [0]);
sam_B = bptr [1] + apply_weight (dpp->weight_B, dpp->samples_B [0]);
update_weight_clip (dpp->weight_B, dpp->delta, dpp->samples_B [0], bptr [1]);
bptr [0] = dpp->samples_B [0] = sam_A;
bptr [1] = dpp->samples_A [0] = sam_B;
}
break;
}
}
#ifdef FAST_DECODE
// This function is a specialized version of decorr_stereo_pass() that works
// only with lower resolution data (<= 16-bit), but is otherwise identical.
static void decorr_stereo_pass_i (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t *bptr, *eptr = buffer + (sample_count * 2);
int m, k;
switch (dpp->term) {
case 17:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam, tmp;
sam = 2 * dpp->samples_A [0] - dpp->samples_A [1];
dpp->samples_A [1] = dpp->samples_A [0];
bptr [0] = dpp->samples_A [0] = apply_weight_i (dpp->weight_A, sam) + (tmp = bptr [0]);
update_weight (dpp->weight_A, dpp->delta, sam, tmp);
sam = 2 * dpp->samples_B [0] - dpp->samples_B [1];
dpp->samples_B [1] = dpp->samples_B [0];
bptr [1] = dpp->samples_B [0] = apply_weight_i (dpp->weight_B, sam) + (tmp = bptr [1]);
update_weight (dpp->weight_B, dpp->delta, sam, tmp);
}
break;
case 18:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam, tmp;
sam = dpp->samples_A [0] + ((dpp->samples_A [0] - dpp->samples_A [1]) >> 1);
dpp->samples_A [1] = dpp->samples_A [0];
bptr [0] = dpp->samples_A [0] = apply_weight_i (dpp->weight_A, sam) + (tmp = bptr [0]);
update_weight (dpp->weight_A, dpp->delta, sam, tmp);
sam = dpp->samples_B [0] + ((dpp->samples_B [0] - dpp->samples_B [1]) >> 1);
dpp->samples_B [1] = dpp->samples_B [0];
bptr [1] = dpp->samples_B [0] = apply_weight_i (dpp->weight_B, sam) + (tmp = bptr [1]);
update_weight (dpp->weight_B, dpp->delta, sam, tmp);
}
break;
default:
for (m = 0, k = dpp->term & (MAX_TERM - 1), bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = dpp->samples_A [m];
dpp->samples_A [k] = apply_weight_i (dpp->weight_A, sam) + bptr [0];
update_weight (dpp->weight_A, dpp->delta, sam, bptr [0]);
bptr [0] = dpp->samples_A [k];
sam = dpp->samples_B [m];
dpp->samples_B [k] = apply_weight_i (dpp->weight_B, sam) + bptr [1];
update_weight (dpp->weight_B, dpp->delta, sam, bptr [1]);
bptr [1] = dpp->samples_B [k];
m = (m + 1) & (MAX_TERM - 1);
k = (k + 1) & (MAX_TERM - 1);
}
break;
case -1:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = bptr [0] + apply_weight_i (dpp->weight_A, dpp->samples_A [0]);
update_weight_clip (dpp->weight_A, dpp->delta, dpp->samples_A [0], bptr [0]);
bptr [0] = sam;
dpp->samples_A [0] = bptr [1] + apply_weight_i (dpp->weight_B, sam);
update_weight_clip (dpp->weight_B, dpp->delta, sam, bptr [1]);
bptr [1] = dpp->samples_A [0];
}
break;
case -2:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = bptr [1] + apply_weight_i (dpp->weight_B, dpp->samples_B [0]);
update_weight_clip (dpp->weight_B, dpp->delta, dpp->samples_B [0], bptr [1]);
bptr [1] = sam;
dpp->samples_B [0] = bptr [0] + apply_weight_i (dpp->weight_A, sam);
update_weight_clip (dpp->weight_A, dpp->delta, sam, bptr [0]);
bptr [0] = dpp->samples_B [0];
}
break;
case -3:
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam_A, sam_B;
sam_A = bptr [0] + apply_weight_i (dpp->weight_A, dpp->samples_A [0]);
update_weight_clip (dpp->weight_A, dpp->delta, dpp->samples_A [0], bptr [0]);
sam_B = bptr [1] + apply_weight_i (dpp->weight_B, dpp->samples_B [0]);
update_weight_clip (dpp->weight_B, dpp->delta, dpp->samples_B [0], bptr [1]);
bptr [0] = dpp->samples_B [0] = sam_A;
bptr [1] = dpp->samples_A [0] = sam_B;
}
break;
}
}
// These functions are specialized versions of decorr_stereo_pass() that work
// only with lower resolution data (<= 16-bit) and handle the equivalent of
// *two* decorrelation passes. By combining two passes we save a read and write
// of the sample data and some overhead dealing with buffer pointers and looping.
//
// The cases handled are:
// 17,17 -- standard "fast" mode before version 4.40
// 17,18 -- standard "fast" mode starting with 4.40
// 18,18 -- used in the default and higher modes
// [1-7],[1-7] -- common in "high" and "very high" modes
static void decorr_stereo_pass_1718 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t *bptr, *eptr = buffer + (sample_count * 2);
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = 2 * dpp->samples_A [0] - dpp->samples_A [1];
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_A [0] = apply_weight_i (dpp->weight_A, sam) + bptr [0];
update_weight (dpp->weight_A, dpp->delta, sam, bptr [0]);
sam = (dpp+1)->samples_A [0] + (((dpp+1)->samples_A [0] - (dpp+1)->samples_A [1]) >> 1);
(dpp+1)->samples_A [1] = (dpp+1)->samples_A [0];
bptr [0] = (dpp+1)->samples_A [0] = apply_weight_i ((dpp+1)->weight_A, sam) + dpp->samples_A [0];
update_weight ((dpp+1)->weight_A, (dpp+1)->delta, sam, dpp->samples_A [0]);
sam = 2 * dpp->samples_B [0] - dpp->samples_B [1];
dpp->samples_B [1] = dpp->samples_B [0];
dpp->samples_B [0] = apply_weight_i (dpp->weight_B, sam) + bptr [1];
update_weight (dpp->weight_B, dpp->delta, sam, bptr [1]);
sam = (dpp+1)->samples_B [0] + (((dpp+1)->samples_B [0] - (dpp+1)->samples_B [1]) >> 1);
(dpp+1)->samples_B [1] = (dpp+1)->samples_B [0];
bptr [1] = (dpp+1)->samples_B [0] = apply_weight_i ((dpp+1)->weight_B, sam) + dpp->samples_B [0];
update_weight ((dpp+1)->weight_B, (dpp+1)->delta, sam, dpp->samples_B [0]);
}
}
static void decorr_stereo_pass_1717 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t *bptr, *eptr = buffer + (sample_count * 2);
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = 2 * dpp->samples_A [0] - dpp->samples_A [1];
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_A [0] = apply_weight_i (dpp->weight_A, sam) + bptr [0];
update_weight (dpp->weight_A, dpp->delta, sam, bptr [0]);
sam = 2 * (dpp+1)->samples_A [0] - (dpp+1)->samples_A [1];
(dpp+1)->samples_A [1] = (dpp+1)->samples_A [0];
bptr [0] = (dpp+1)->samples_A [0] = apply_weight_i ((dpp+1)->weight_A, sam) + dpp->samples_A [0];
update_weight ((dpp+1)->weight_A, (dpp+1)->delta, sam, dpp->samples_A [0]);
sam = 2 * dpp->samples_B [0] - dpp->samples_B [1];
dpp->samples_B [1] = dpp->samples_B [0];
dpp->samples_B [0] = apply_weight_i (dpp->weight_B, sam) + bptr [1];
update_weight (dpp->weight_B, dpp->delta, sam, bptr [1]);
sam = 2 * (dpp+1)->samples_B [0] - (dpp+1)->samples_B [1];
(dpp+1)->samples_B [1] = (dpp+1)->samples_B [0];
bptr [1] = (dpp+1)->samples_B [0] = apply_weight_i ((dpp+1)->weight_B, sam) + dpp->samples_B [0];
update_weight ((dpp+1)->weight_B, (dpp+1)->delta, sam, dpp->samples_B [0]);
}
}
static void decorr_stereo_pass_1818 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t *bptr, *eptr = buffer + (sample_count * 2);
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t sam;
sam = dpp->samples_A [0] + ((dpp->samples_A [0] - dpp->samples_A [1]) >> 1);
dpp->samples_A [1] = dpp->samples_A [0];
dpp->samples_A [0] = apply_weight_i (dpp->weight_A, sam) + bptr [0];
update_weight (dpp->weight_A, dpp->delta, sam, bptr [0]);
sam = (dpp+1)->samples_A [0] + (((dpp+1)->samples_A [0] - (dpp+1)->samples_A [1]) >> 1);
(dpp+1)->samples_A [1] = (dpp+1)->samples_A [0];
bptr [0] = (dpp+1)->samples_A [0] = apply_weight_i ((dpp+1)->weight_A, sam) + dpp->samples_A [0];
update_weight ((dpp+1)->weight_A, (dpp+1)->delta, sam, dpp->samples_A [0]);
sam = dpp->samples_B [0] + ((dpp->samples_B [0] - dpp->samples_B [1]) >> 1);
dpp->samples_B [1] = dpp->samples_B [0];
dpp->samples_B [0] = apply_weight_i (dpp->weight_B, sam) + bptr [1];
update_weight (dpp->weight_B, dpp->delta, sam, bptr [1]);
sam = (dpp+1)->samples_B [0] + (((dpp+1)->samples_B [0] - (dpp+1)->samples_B [1]) >> 1);
(dpp+1)->samples_B [1] = (dpp+1)->samples_B [0];
bptr [1] = (dpp+1)->samples_B [0] = apply_weight_i ((dpp+1)->weight_B, sam) + dpp->samples_B [0];
update_weight ((dpp+1)->weight_B, (dpp+1)->delta, sam, dpp->samples_B [0]);
}
}
static void decorr_stereo_pass_nn (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
{
int32_t *bptr, *eptr = buffer + (sample_count * 2);
int m, k, j;
m = 0;
k = dpp->term & (MAX_TERM - 1);
j = (dpp+1)->term & (MAX_TERM - 1);
for (bptr = buffer; bptr < eptr; bptr += 2) {
int32_t tmp;
dpp->samples_A [k] = apply_weight_i (dpp->weight_A, dpp->samples_A [m]) + (tmp = bptr [0]);
update_weight (dpp->weight_A, dpp->delta, dpp->samples_A [m], tmp);
bptr [0] = (dpp+1)->samples_A [j] = apply_weight_i ((dpp+1)->weight_A, (dpp+1)->samples_A [m]) + (tmp = dpp->samples_A [k]);
update_weight ((dpp+1)->weight_A, (dpp+1)->delta, (dpp+1)->samples_A [m], tmp);
dpp->samples_B [k] = apply_weight_i (dpp->weight_B, dpp->samples_B [m]) + (tmp = bptr [1]);
update_weight (dpp->weight_B, dpp->delta, dpp->samples_B [m], tmp);
bptr [1] = (dpp+1)->samples_B [j] = apply_weight_i ((dpp+1)->weight_B, (dpp+1)->samples_B [m]) + (tmp = dpp->samples_B [k]);
update_weight ((dpp+1)->weight_B, (dpp+1)->delta, (dpp+1)->samples_B [m], tmp);
m = (m + 1) & (MAX_TERM - 1);
k = (k + 1) & (MAX_TERM - 1);
j = (j + 1) & (MAX_TERM - 1);
}
}
#endif
// 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 (WavpackContext *wpc, int32_t *buffer, uint32_t sample_count)
{
WavpackStream *wps = wpc->streams [wpc->current_stream];
uint32_t flags = wps->wphdr.flags;
int lossy_flag = (flags & HYBRID_FLAG) && !wps->block2buff;
int shift = (flags & SHIFT_MASK) >> SHIFT_LSB;
if (flags & FLOAT_DATA) {
float_values (wps, buffer, (flags & MONO_DATA) ? sample_count : sample_count * 2);
return;
}
if (flags & INT32_DATA) {
uint32_t count = (flags & MONO_DATA) ? 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;
uint32_t data, mask = (1 << sent_bits) - 1;
int32_t *dptr = buffer;
if (bs_is_open (&wps->wvxbits)) {
uint32_t crc = wps->crc_x;
while (count--) {
// if (sent_bits) {
getbits (&data, sent_bits, &wps->wvxbits);
*dptr = (*dptr << sent_bits) | (data & mask);
// }
if (zeros)
*dptr <<= zeros;
else if (ones)
*dptr = ((*dptr + 1) << ones) - 1;
else if (dups)
*dptr = ((*dptr + (*dptr & 1)) << dups) - (*dptr & 1);
crc = crc * 9 + (*dptr & 0xffff) * 3 + ((*dptr >> 16) & 0xffff);
dptr++;
}
wps->crc_x = crc;
}
else if (!sent_bits && (zeros + ones + dups)) {
while (lossy_flag && (flags & BYTES_STORED) == 3 && shift < 8) {
if (zeros)
zeros--;
else if (ones)
ones--;
else if (dups)
dups--;
else
break;
shift++;
}
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 (lossy_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: /* "default" suppresses compiler warning */
min_shifted = (min_value = (int32_t) 0x80000000 >> shift) << shift;
max_shifted = (max_value = (int32_t) 0x7fffffff >> shift) << shift;
break;
}
if (!(flags & MONO_DATA))
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_DATA))
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)
{
int result = 0, stream;
for (stream = 0; stream < wpc->num_streams; stream++) {
WavpackStream *wps = wpc->streams [stream];
if (wps->crc != wps->wphdr.crc)
++result;
else if (bs_is_open (&wps->wvxbits) && wps->crc_x != wps->crc_wvx)
++result;
}
return result;
}
#endif