cog/Frameworks/WavPack/Files/unpack.c

818 lines
31 KiB
C

////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2013 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 read_words.c module. For better
// efficiency, the conversion is isolated to tight loops that handle an entire
// buffer.
#include <stdlib.h>
#include <string.h>
#include "wavpack_local.h"
#ifdef OPT_ASM_X86
#define DECORR_STEREO_PASS_CONT unpack_decorr_stereo_pass_cont_x86
#define DECORR_STEREO_PASS_CONT_AVAILABLE unpack_cpu_has_feature_x86(CPU_FEATURE_MMX)
#define DECORR_MONO_PASS_CONT unpack_decorr_mono_pass_cont_x86
#elif defined(OPT_ASM_X64) && (defined (_WIN64) || defined(__CYGWIN__) || defined(__MINGW64__) || defined(__midipix__))
#define DECORR_STEREO_PASS_CONT unpack_decorr_stereo_pass_cont_x64win
#define DECORR_STEREO_PASS_CONT_AVAILABLE 1
#define DECORR_MONO_PASS_CONT unpack_decorr_mono_pass_cont_x64win
#elif defined(OPT_ASM_X64)
#define DECORR_STEREO_PASS_CONT unpack_decorr_stereo_pass_cont_x64
#define DECORR_STEREO_PASS_CONT_AVAILABLE 1
#define DECORR_MONO_PASS_CONT unpack_decorr_mono_pass_cont_x64
#elif defined(OPT_ASM_ARM)
#define DECORR_STEREO_PASS_CONT unpack_decorr_stereo_pass_cont_armv7
#define DECORR_STEREO_PASS_CONT_AVAILABLE 1
#define DECORR_MONO_PASS_CONT unpack_decorr_mono_pass_cont_armv7
#endif
#ifdef DECORR_STEREO_PASS_CONT
extern void DECORR_STEREO_PASS_CONT (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count, int32_t long_math);
extern void DECORR_MONO_PASS_CONT (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count, int32_t long_math);
#endif
// This flag provides the functionality of terminating the decoding and muting
// the output when a lossy sample appears to be corrupt. This is automatic
// for lossless files because a corrupt sample is unambigious, but for lossy
// data it might be possible for this to falsely trigger (although I have never
// seen it).
#define LOSSY_MUTE
///////////////////////////// executable code ////////////////////////////////
// This monster actually unpacks the WavPack bitstream(s) into the specified
// buffer as 32-bit integers or floats (depending on original 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_mono_pass (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 = (1L << ((flags & MAG_MASK) >> MAG_LSB)) + 2;
int32_t correction [2], read_word, *bptr;
struct decorr_pass *dpp;
int tcount, m = 0;
// don't attempt to decode past the end of the block, but watch out for overflow!
if (wps->sample_index + sample_count > GET_BLOCK_INDEX (wps->wphdr) + wps->wphdr.block_samples &&
(uint32_t) (GET_BLOCK_INDEX (wps->wphdr) + wps->wphdr.block_samples - wps->sample_index) < sample_count)
sample_count = (uint32_t) (GET_BLOCK_INDEX (wps->wphdr) + wps->wphdr.block_samples - wps->sample_index);
if (GET_BLOCK_INDEX (wps->wphdr) > wps->sample_index || wps->wphdr.block_samples < sample_count)
wps->mute_error = TRUE;
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 = (mute_limit * 2) + 128;
//////////////// 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);
#ifdef DECORR_MONO_PASS_CONT
if (sample_count < 16)
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
decorr_mono_pass (dpp, buffer, sample_count);
else
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
int pre_samples = (dpp->term > MAX_TERM) ? 2 : dpp->term;
decorr_mono_pass (dpp, buffer, pre_samples);
DECORR_MONO_PASS_CONT (dpp, buffer + pre_samples, sample_count - pre_samples,
((flags & MAG_MASK) >> MAG_LSB) > 15);
}
#else
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
decorr_mono_pass (dpp, buffer, sample_count);
#endif
#ifndef LOSSY_MUTE
if (!(flags & HYBRID_FLAG))
#endif
for (bptr = buffer; bptr < eptr; ++bptr) {
if (labs (bptr [0]) > mute_limit) {
i = (uint32_t)(bptr - buffer);
break;
}
crc = crc * 3 + bptr [0];
}
#ifndef LOSSY_MUTE
else
for (bptr = buffer; bptr < eptr; ++bptr)
crc = crc * 3 + bptr [0];
#endif
}
/////////////// 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 DECORR_STEREO_PASS_CONT
if (sample_count < 16 || !DECORR_STEREO_PASS_CONT_AVAILABLE) {
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
decorr_stereo_pass (dpp, buffer, sample_count);
m = sample_count & (MAX_TERM - 1);
}
else
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
int pre_samples = (dpp->term < 0 || dpp->term > MAX_TERM) ? 2 : dpp->term;
decorr_stereo_pass (dpp, buffer, pre_samples);
DECORR_STEREO_PASS_CONT (dpp, buffer + pre_samples * 2, sample_count - pre_samples,
((flags & MAG_MASK) >> MAG_LSB) >= 16);
}
#else
for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
decorr_stereo_pass (dpp, buffer, sample_count);
m = sample_count & (MAX_TERM - 1);
#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];
#ifndef LOSSY_MUTE
if (!(flags & HYBRID_FLAG))
#endif
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;
}
}
/////////////////// 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;
if (labs (read_word) > mute_limit)
break;
*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;
}
if (labs (left) > mute_limit || labs (right) > mute_limit)
break;
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 mono 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 returned normalized for term
// values 1-8.
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;
}
// 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;
}
}
// 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;
}