////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2013 Conifer Software. //
// All Rights Reserved. //
// Distributed under the BSD Software License (see license.txt) //
////////////////////////////////////////////////////////////////////////////
// pack_utils.c
// This module provides the high-level API for creating WavPack files from
// audio data. It manages the buffers used to deinterleave the data passed
// in from the application into the individual streams and it handles the
// generation of riff headers and the "fixup" on the first WavPack block
// header for the case where the number of samples was unknown (or wrong).
// The actual audio stream compression is handled in the pack.c module.
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "wavpack_local.h"
///////////////////////////// executable code ////////////////////////////////
// Open context for writing WavPack files. The returned context pointer is used
// in all following calls to the library. The "blockout" function will be used
// to store the actual completed WavPack blocks and will be called with the id
// pointers containing user defined data (one for the wv file and one for the
// wvc file). A return value of NULL indicates that memory could not be
// allocated for the context.
WavpackContext *WavpackOpenFileOutput (WavpackBlockOutput blockout, void *wv_id, void *wvc_id)
{
WavpackContext *wpc = malloc (sizeof (WavpackContext));
if (!wpc)
return NULL;
CLEAR (*wpc);
wpc->total_samples = -1;
wpc->stream_version = CUR_STREAM_VERS;
wpc->blockout = blockout;
wpc->wv_out = wv_id;
wpc->wvc_out = wvc_id;
return wpc;
}
static int add_to_metadata (WavpackContext *wpc, void *data, uint32_t bcount, unsigned char id);
// New for version 5.0, this function allows the application to store a file extension and a
// file_format identification. The extension would be used by the unpacker if the user had not
// specified the target filename, and specifically handles the case where the original file
// had the "wrong" extension for the file format (e.g., a Wave64 file having a "wav" extension)
// or an alternative (e.g., "bwf") or where the file format is not known. Specifying a file
// format besides the default WP_FORMAT_WAV will ensure that old decoders will not be able to
// see the non-wav wrapper provided with WavpackAddWrapper() (which they would end up putting
// on a file with a .wav extension).
void WavpackSetFileInformation (WavpackContext *wpc, char *file_extension, unsigned char file_format)
{
if (file_extension && strlen (file_extension) < sizeof (wpc->file_extension)) {
add_to_metadata (wpc, file_extension, (uint32_t) strlen (file_extension), ID_ALT_EXTENSION);
strcpy (wpc->file_extension, file_extension);
}
wpc->file_format = file_format;
}
// Set configuration for writing WavPack files. This must be done before
// sending any actual samples, however it is okay to send wrapper or other
// metadata before calling this. The "config" structure contains the following
// required information:
// config->bytes_per_sample see WavpackGetBytesPerSample() for info
// config->bits_per_sample see WavpackGetBitsPerSample() for info
// config->channel_mask Microsoft standard (mono = 4, stereo = 3)
// config->num_channels self evident
// config->sample_rate self evident
// In addition, the following fields and flags may be set:
// config->flags:
// --------------
// o CONFIG_HYBRID_FLAG select hybrid mode (must set bitrate)
// o CONFIG_JOINT_STEREO select joint stereo (must set override also)
// o CONFIG_JOINT_OVERRIDE override default joint stereo selection
// o CONFIG_HYBRID_SHAPE select hybrid noise shaping (set override &
// shaping_weight != 0.0)
// o CONFIG_SHAPE_OVERRIDE override default hybrid noise shaping
// (set CONFIG_HYBRID_SHAPE and shaping_weight)
// o CONFIG_FAST_FLAG "fast" compression mode
// o CONFIG_HIGH_FLAG "high" compression mode
// o CONFIG_BITRATE_KBPS hybrid bitrate is kbps, not bits / sample
// o CONFIG_CREATE_WVC create correction file
// o CONFIG_OPTIMIZE_WVC maximize bybrid compression (-cc option)
// o CONFIG_CALC_NOISE calc noise in hybrid mode
// o CONFIG_EXTRA_MODE extra processing mode (slow!)
// o CONFIG_SKIP_WVX no wvx stream for floats & large ints
// o CONFIG_MD5_CHECKSUM specify if you plan to store MD5 signature
// o CONFIG_CREATE_EXE specify if you plan to prepend sfx module
// o CONFIG_OPTIMIZE_MONO detect and optimize for mono files posing as
// stereo (uses a more recent stream format that
// is not compatible with decoders < 4.3)
// config->bitrate hybrid bitrate in either bits/sample or kbps
// config->shaping_weight hybrid noise shaping coefficient override
// config->block_samples force samples per WavPack block (0 = use deflt)
// config->float_norm_exp select floating-point data (127 for +/-1.0)
// config->xmode extra mode processing value override
// If the number of samples to be written is known then it should be passed
// here. If the duration is not known then pass -1. In the case that the size
// is not known (or the writing is terminated early) then it is suggested that
// the application retrieve the first block written and let the library update
// the total samples indication. A function is provided to do this update and
// it should be done to the "correction" file also. If this cannot be done
// (because a pipe is being used, for instance) then a valid WavPack will still
// be created, but when applications want to access that file they will have
// to seek all the way to the end to determine the actual duration. Also, if
// a RIFF header has been included then it should be updated as well or the
// WavPack file will not be directly unpackable to a valid wav file (although
// it will still be usable by itself). A return of FALSE indicates an error.
//
// The enhanced version of this function now allows setting the identities of
// any channels that are NOT standard Microsoft channels and are therefore not
// represented in the channel mask. WavPack files require that all the Microsoft
// channels come first (and in Microsoft order) and these are followed by any
// other channels (which can be in any order).
//
// The identities are provided in a NULL-terminated string (0x00 is not an allowed
// channel ID). The Microsoft channels may be provided as well (and will be checked)
// but it is really only necessary to provide the "unknown" channels. Any truly
// unknown channels are indicated with a 0xFF.
//
// The channel IDs so far reserved are listed here:
//
// 0: not allowed / terminator
// 1 - 18: Microsoft standard channels
// 30, 31: Stereo mix from RF64 (not really recommended, but RF64 specifies this)
// 33 - 44: Core Audio channels (see Core Audio specification)
// 127 - 128: Amio LeftHeight, Amio RightHeight
// 138 - 142: Amio BottomFrontLeft/Center/Right, Amio ProximityLeft/Right
// 200 - 207: Core Audio channels (see Core Audio specification)
// 221 - 224: Core Audio channels 301 - 305 (offset by 80)
// 255: Present but unknown or unused channel
//
// All other channel IDs are reserved. Ask if something you need is missing.
// Table of channels that will automatically "pair" into a single stereo stream
static const struct { unsigned char a, b; } stereo_pairs [] = {
{ 1, 2 }, // FL, FR
{ 5, 6 }, // BL, BR
{ 7, 8 }, // FLC, FRC
{ 10, 11 }, // SL, SR
{ 13, 15 }, // TFL, TFR
{ 16, 18 }, // TBL, TBR
{ 30, 31 }, // stereo mix L,R (RF64)
{ 33, 34 }, // Rls, Rrs
{ 35, 36 }, // Lw, Rw
{ 38, 39 }, // Lt, Rt
{ 127, 128 }, // Lh, Rh
{ 138, 140 }, // Bfl, Bfr
{ 141, 142 }, // Pl, Pr
{ 200, 201 }, // Amb_W, Amb_X
{ 202, 203 }, // Amb_Y, Amb_Z
{ 204, 205 }, // MS_Mid, MS_Side
{ 206, 207 }, // XY_X, XY_Y
{ 221, 222 }, // Hph_L, Hph_R
};
#define NUM_STEREO_PAIRS (sizeof (stereo_pairs) / sizeof (stereo_pairs [0]))
// Legacy version of this function for compatibility with existing applications. Note that this version
// also generates older streams to be compatible with all decoders back to 4.0, but of course cannot be
// used with > 2^32 samples or non-Microsoft channels. The older stream version only differs in that it
// does not support the "mono optimization" feature where stereo blocks containing identical audio data
// in both channels are encoded in mono for better efficiency.
int WavpackSetConfiguration (WavpackContext *wpc, WavpackConfig *config, uint32_t total_samples)
{
config->flags |= CONFIG_COMPATIBLE_WRITE; // write earlier version streams
if (total_samples == (uint32_t) -1)
return WavpackSetConfiguration64 (wpc, config, -1, NULL);
else
return WavpackSetConfiguration64 (wpc, config, total_samples, NULL);
}
int WavpackSetConfiguration64 (WavpackContext *wpc, WavpackConfig *config, int64_t total_samples, const unsigned char *chan_ids)
{
uint32_t flags, bps = 0;
uint32_t chan_mask = config->channel_mask;
int num_chans = config->num_channels;
int i;
if (!config->sample_rate) {
strcpy (wpc->error_message, "sample rate cannot be zero!");
return FALSE;
}
if (!num_chans) {
strcpy (wpc->error_message, "channel count cannot be zero!");
return FALSE;
}
wpc->stream_version = (config->flags & CONFIG_COMPATIBLE_WRITE) ? CUR_STREAM_VERS : MAX_STREAM_VERS;
if ((config->qmode & QMODE_DSD_AUDIO) && config->bytes_per_sample == 1 && config->bits_per_sample == 8) {
#ifdef ENABLE_DSD
wpc->dsd_multiplier = 1;
flags = DSD_FLAG;
for (i = 14; i >= 0; --i)
if (config->sample_rate % sample_rates [i] == 0) {
int divisor = config->sample_rate / sample_rates [i];
if (divisor && (divisor & (divisor - 1)) == 0) {
config->sample_rate /= divisor;
wpc->dsd_multiplier = divisor;
break;
}
}
// most options that don't apply to DSD we can simply ignore for now, but NOT hybrid mode!
if (config->flags & CONFIG_HYBRID_FLAG) {
strcpy (wpc->error_message, "hybrid mode not available for DSD!");
return FALSE;
}
// with DSD, very few PCM options work (or make sense), so only allow those that do
config->flags &= (CONFIG_HIGH_FLAG | CONFIG_MD5_CHECKSUM | CONFIG_PAIR_UNDEF_CHANS);
config->float_norm_exp = config->xmode = 0;
#else
strcpy (wpc->error_message, "libwavpack not configured for DSD!");
return FALSE;
#endif
}
else
flags = config->bytes_per_sample - 1;
wpc->total_samples = total_samples;
wpc->config.sample_rate = config->sample_rate;
wpc->config.num_channels = config->num_channels;
wpc->config.channel_mask = config->channel_mask;
wpc->config.bits_per_sample = config->bits_per_sample;
wpc->config.bytes_per_sample = config->bytes_per_sample;
wpc->config.block_samples = config->block_samples;
wpc->config.flags = config->flags;
wpc->config.qmode = config->qmode;
if (config->flags & CONFIG_VERY_HIGH_FLAG)
wpc->config.flags |= CONFIG_HIGH_FLAG;
for (i = 0; i < 15; ++i)
if (wpc->config.sample_rate == sample_rates [i])
break;
flags |= i << SRATE_LSB;
// all of this stuff only applies to PCM
if (!(flags & DSD_FLAG)) {
if (config->float_norm_exp) {
if (config->bytes_per_sample != 4 || config->bits_per_sample != 32) {
strcpy (wpc->error_message, "incorrect bits/bytes configuration for float data!");
return FALSE;
}
wpc->config.float_norm_exp = config->float_norm_exp;
wpc->config.flags |= CONFIG_FLOAT_DATA;
flags |= FLOAT_DATA;
}
else {
if (config->bytes_per_sample < 1 || config->bytes_per_sample > 4) {
strcpy (wpc->error_message, "invalid bytes per sample!");
return FALSE;
}
if (config->bits_per_sample < 1 || config->bits_per_sample > config->bytes_per_sample * 8) {
strcpy (wpc->error_message, "invalid bits per sample!");
return FALSE;
}
flags |= ((config->bytes_per_sample * 8) - config->bits_per_sample) << SHIFT_LSB;
}
if (config->flags & CONFIG_HYBRID_FLAG) {
flags |= HYBRID_FLAG | HYBRID_BITRATE | HYBRID_BALANCE;
if (!(wpc->config.flags & CONFIG_SHAPE_OVERRIDE)) {
wpc->config.flags |= CONFIG_HYBRID_SHAPE | CONFIG_AUTO_SHAPING;
flags |= HYBRID_SHAPE | NEW_SHAPING;
}
else if (wpc->config.flags & CONFIG_HYBRID_SHAPE) {
wpc->config.shaping_weight = config->shaping_weight;
flags |= HYBRID_SHAPE | NEW_SHAPING;
}
if (wpc->config.flags & (CONFIG_CROSS_DECORR | CONFIG_OPTIMIZE_WVC))
flags |= CROSS_DECORR;
if (config->flags & CONFIG_BITRATE_KBPS) {
bps = (uint32_t) floor (config->bitrate * 256000.0 / config->sample_rate / config->num_channels + 0.5);
if (bps > (64 << 8))
bps = 64 << 8;
}
else
bps = (uint32_t) floor (config->bitrate * 256.0 + 0.5);
}
else
flags |= CROSS_DECORR;
if (!(config->flags & CONFIG_JOINT_OVERRIDE) || (config->flags & CONFIG_JOINT_STEREO))
flags |= JOINT_STEREO;
if (config->flags & CONFIG_CREATE_WVC)
wpc->wvc_flag = TRUE;
}
// if a channel-identities string was specified, process that here, otherwise all channels
// not present in the channel mask are considered "unassigned"
if (chan_ids) {
int lastchan = 0, mask_copy = chan_mask;
if ((int) strlen ((char *) chan_ids) > num_chans) { // can't be more than num channels!
strcpy (wpc->error_message, "chan_ids longer than num channels!");
return FALSE;
}
// skip past channels that are specified in the channel mask (no reason to store those)
while (*chan_ids)
if (*chan_ids <= 32 && *chan_ids > lastchan && (mask_copy & (1 << (*chan_ids-1)))) {
mask_copy &= ~(1 << (*chan_ids-1));
lastchan = *chan_ids++;
}
else
break;
// now scan the string for an actually defined channel (and don't store if there aren't any)
for (i = 0; chan_ids [i]; i++)
if (chan_ids [i] != 0xff) {
wpc->channel_identities = (unsigned char *) strdup ((char *) chan_ids);
break;
}
}
// This loop goes through all the channels and creates the Wavpack "streams" for them to go in.
// A stream can hold either one or two channels, so we have several rules to determine how many
// channels will go in each stream.
for (wpc->current_stream = 0; num_chans; wpc->current_stream++) {
WavpackStream *wps = malloc (sizeof (WavpackStream));
unsigned char left_chan_id = 0, right_chan_id = 0;
int pos, chans = 1;
// allocate the stream and initialize the pointer to it
wpc->streams = realloc (wpc->streams, (wpc->current_stream + 1) * sizeof (wpc->streams [0]));
wpc->streams [wpc->current_stream] = wps;
CLEAR (*wps);
// if there are any bits [still] set in the channel_mask, get the next one or two IDs from there
if (chan_mask)
for (pos = 0; pos < 32; ++pos)
if (chan_mask & (1 << pos)) {
if (left_chan_id) {
right_chan_id = pos + 1;
break;
}
else {
chan_mask &= ~(1 << pos);
left_chan_id = pos + 1;
}
}
// next check for any channels identified in the channel-identities string
while (!right_chan_id && chan_ids && *chan_ids)
if (left_chan_id)
right_chan_id = *chan_ids;
else
left_chan_id = *chan_ids++;
// assume anything we did not get is "unassigned"
if (!left_chan_id)
left_chan_id = right_chan_id = 0xff;
else if (!right_chan_id)
right_chan_id = 0xff;
// if we have 2 channels, this is where we decide if we can combine them into one stream:
// 1. they are "unassigned" and we've been told to combine unassigned pairs, or
// 2. they appear together in the valid "pairings" list
if (num_chans >= 2) {
if ((config->flags & CONFIG_PAIR_UNDEF_CHANS) && left_chan_id == 0xff && right_chan_id == 0xff)
chans = 2;
else
for (i = 0; i < NUM_STEREO_PAIRS; ++i)
if ((left_chan_id == stereo_pairs [i].a && right_chan_id == stereo_pairs [i].b) ||
(left_chan_id == stereo_pairs [i].b && right_chan_id == stereo_pairs [i].a)) {
if (right_chan_id <= 32 && (chan_mask & (1 << (right_chan_id-1))))
chan_mask &= ~(1 << (right_chan_id-1));
else if (chan_ids && *chan_ids == right_chan_id)
chan_ids++;
chans = 2;
break;
}
}
num_chans -= chans;
if (num_chans && wpc->current_stream == NEW_MAX_STREAMS - 1)
break;
memcpy (wps->wphdr.ckID, "wvpk", 4);
wps->wphdr.ckSize = sizeof (WavpackHeader) - 8;
SET_TOTAL_SAMPLES (wps->wphdr, wpc->total_samples);
wps->wphdr.version = wpc->stream_version;
wps->wphdr.flags = flags;
wps->bits = bps;
if (!wpc->current_stream)
wps->wphdr.flags |= INITIAL_BLOCK;
if (!num_chans)
wps->wphdr.flags |= FINAL_BLOCK;
if (chans == 1) {
wps->wphdr.flags &= ~(JOINT_STEREO | CROSS_DECORR | HYBRID_BALANCE);
wps->wphdr.flags |= MONO_FLAG;
}
}
wpc->num_streams = wpc->current_stream;
wpc->current_stream = 0;
if (num_chans) {
strcpy (wpc->error_message, "too many channels!");
return FALSE;
}
if (config->flags & CONFIG_EXTRA_MODE)
wpc->config.xmode = config->xmode ? config->xmode : 1;
return TRUE;
}
// This function allows setting the Core Audio File channel layout, many of which do not
// conform to the Microsoft ordering standard that Wavpack requires internally (at least for
// those channels present in the "channel mask"). In addition to the layout tag, this function
// allows a reordering string to be stored in the file to allow the unpacker to reorder the
// channels back to the specified layout (if it is aware of this feature and wants to restore
// the CAF order). The number of channels in the layout is specified in the lower nybble of
// the layout word, and if a reorder string is specified it must be that long. Note that all
// the reordering is actually done outside of this library, and that if reordering is done
// then the appropriate qmode bit must be set to ensure that any MD5 sum is stored with a new
// ID so that old decoders don't try to verify it (and to let the decoder know that a reorder
// might be required).
//
// Note: This function should only be used to encode Core Audio files in such a way that a
// verbatim archive can be created. Applications can just include the chan_ids parameter in
// the call to WavpackSetConfiguration64() if there are non-Microsoft channels to specify,
// or do nothing special if only Microsoft channels are present (the vast majority of cases).
int WavpackSetChannelLayout (WavpackContext *wpc, uint32_t layout_tag, const unsigned char *reorder)
{
int nchans = layout_tag & 0xff;
if ((layout_tag & 0xff00ff00) || nchans > wpc->config.num_channels)
return FALSE;
wpc->channel_layout = layout_tag;
if (wpc->channel_reordering) {
free (wpc->channel_reordering);
wpc->channel_reordering = NULL;
}
if (nchans && reorder) {
int min_index = 256, i;
for (i = 0; i < nchans; ++i)
if (reorder [i] < min_index)
min_index = reorder [i];
wpc->channel_reordering = malloc (nchans);
if (wpc->channel_reordering)
for (i = 0; i < nchans; ++i)
wpc->channel_reordering [i] = reorder [i] - min_index;
}
return TRUE;
}
// Prepare to actually pack samples by determining the size of the WavPack
// blocks and allocating sample buffers and initializing each stream. Call
// after WavpackSetConfiguration() and before WavpackPackSamples(). A return
// of FALSE indicates an error.
static int write_metadata_block (WavpackContext *wpc);
int WavpackPackInit (WavpackContext *wpc)
{
if (wpc->metabytes > 16384) // 16384 bytes still leaves plenty of room for audio
write_metadata_block (wpc); // in this block (otherwise write a special one)
// The default block size is a compromise. Longer blocks provide better encoding efficiency,
// but longer blocks adversely affect memory requirements and seeking performance. For WavPack
// version 5.0, the default block sizes have been reduced by half from the previous version,
// but the difference in encoding efficiency will generally be less than 0.1 percent.
if (wpc->dsd_multiplier) {
wpc->block_samples = (wpc->config.sample_rate % 7) ? 48000 : 44100;
if (wpc->config.flags & CONFIG_HIGH_FLAG)
wpc->block_samples /= 2;
if (wpc->config.num_channels == 1)
wpc->block_samples *= 2;
while (wpc->block_samples > 12000 && wpc->block_samples * wpc->config.num_channels > 300000)
wpc->block_samples /= 2;
}
else {
int divisor = (wpc->config.flags & CONFIG_HIGH_FLAG) ? 2 : 4;
while (wpc->config.sample_rate % divisor)
divisor--;
wpc->block_samples = wpc->config.sample_rate / divisor;
while (wpc->block_samples > 12000 && wpc->block_samples * wpc->config.num_channels > 75000)
wpc->block_samples /= 2;
while (wpc->block_samples * wpc->config.num_channels < 20000)
wpc->block_samples *= 2;
}
if (wpc->config.block_samples) {
if ((wpc->config.flags & CONFIG_MERGE_BLOCKS) &&
wpc->block_samples > (uint32_t) wpc->config.block_samples) {
wpc->block_boundary = wpc->config.block_samples;
wpc->block_samples /= wpc->config.block_samples;
wpc->block_samples *= wpc->config.block_samples;
}
else
wpc->block_samples = wpc->config.block_samples;
}
wpc->ave_block_samples = wpc->block_samples;
wpc->max_samples = wpc->block_samples + (wpc->block_samples >> 1);
for (wpc->current_stream = 0; wpc->current_stream < wpc->num_streams; wpc->current_stream++) {
WavpackStream *wps = wpc->streams [wpc->current_stream];
wps->sample_buffer = malloc (wpc->max_samples * (wps->wphdr.flags & MONO_FLAG ? 4 : 8));
#ifdef ENABLE_DSD
if (wps->wphdr.flags & DSD_FLAG)
pack_dsd_init (wpc);
else
#endif
pack_init (wpc);
}
return TRUE;
}
// Pack the specified samples. Samples must be stored in longs in the native
// endian format of the executing processor. The number of samples specified
// indicates composite samples (sometimes called "frames"). So, the actual
// number of data points would be this "sample_count" times the number of
// channels. Note that samples are accumulated here until enough exist to
// create a complete WavPack block (or several blocks for multichannel audio).
// If an application wants to break a block at a specific sample, then it must
// simply call WavpackFlushSamples() to force an early termination. Completed
// WavPack blocks are send to the function provided in the initial call to
// WavpackOpenFileOutput(). A return of FALSE indicates an error.
static int pack_streams (WavpackContext *wpc, uint32_t block_samples);
static int create_riff_header (WavpackContext *wpc, int64_t total_samples, void *outbuffer);
int WavpackPackSamples (WavpackContext *wpc, int32_t *sample_buffer, uint32_t sample_count)
{
int nch = wpc->config.num_channels;
while (sample_count) {
int32_t *source_pointer = sample_buffer;
unsigned int samples_to_copy;
if (!wpc->riff_header_added && !wpc->riff_header_created && !wpc->file_format) {
char riff_header [128];
if (!add_to_metadata (wpc, riff_header, create_riff_header (wpc, wpc->total_samples, riff_header), ID_RIFF_HEADER))
return FALSE;
}
if (wpc->acc_samples + sample_count > wpc->max_samples)
samples_to_copy = wpc->max_samples - wpc->acc_samples;
else
samples_to_copy = sample_count;
for (wpc->current_stream = 0; wpc->current_stream < wpc->num_streams; wpc->current_stream++) {
WavpackStream *wps = wpc->streams [wpc->current_stream];
int32_t *dptr, *sptr, cnt;
dptr = wps->sample_buffer + wpc->acc_samples * (wps->wphdr.flags & MONO_FLAG ? 1 : 2);
sptr = source_pointer;
cnt = samples_to_copy;
// This code used to just copy the 32-bit samples regardless of the actual size with the
// assumption that the caller had properly sign-extended the values (if they were smaller
// than 32 bits). However, several people have discovered that if the data isn't properly
// sign extended then ugly things happen (e.g. CRC errors that show up only on decode).
// To prevent this, we now explicitly sign-extend samples smaller than 32-bit when we
// copy, and the performance hit from doing this is very small (generally < 1%).
if (wps->wphdr.flags & MONO_FLAG) {
switch (wpc->config.bytes_per_sample) {
case 1:
while (cnt--) {
*dptr++ = (signed char) *sptr;
sptr += nch;
}
break;
case 2:
while (cnt--) {
*dptr++ = (int16_t) *sptr;
sptr += nch;
}
break;
case 3:
while (cnt--) {
*dptr++ = (*sptr << 8) >> 8;
sptr += nch;
}
break;
default:
while (cnt--) {
*dptr++ = *sptr;
sptr += nch;
}
}
source_pointer++;
}
else {
switch (wpc->config.bytes_per_sample) {
case 1:
while (cnt--) {
*dptr++ = (signed char) sptr [0];
*dptr++ = (signed char) sptr [1];
sptr += nch;
}
break;
case 2:
while (cnt--) {
*dptr++ = (int16_t) sptr [0];
*dptr++ = (int16_t) sptr [1];
sptr += nch;
}
break;
case 3:
while (cnt--) {
*dptr++ = (sptr [0] << 8) >> 8;
*dptr++ = (sptr [1] << 8) >> 8;
sptr += nch;
}
break;
default:
while (cnt--) {
*dptr++ = sptr [0];
*dptr++ = sptr [1];
sptr += nch;
}
}
source_pointer += 2;
}
}
sample_buffer += samples_to_copy * nch;
sample_count -= samples_to_copy;
if ((wpc->acc_samples += samples_to_copy) == wpc->max_samples &&
!pack_streams (wpc, wpc->block_samples))
return FALSE;
}
return TRUE;
}
// Flush all accumulated samples into WavPack blocks. This is normally called
// after all samples have been sent to WavpackPackSamples(), but can also be
// called to terminate a WavPack block at a specific sample (in other words it
// is possible to continue after this operation). This is also called to
// dump non-audio blocks like those holding metadata for various purposes.
// A return of FALSE indicates an error.
int WavpackFlushSamples (WavpackContext *wpc)
{
while (wpc->acc_samples) {
uint32_t block_samples;
if (wpc->acc_samples > wpc->block_samples)
block_samples = wpc->acc_samples / 2;
else
block_samples = wpc->acc_samples;
if (!pack_streams (wpc, block_samples))
return FALSE;
}
if (wpc->metacount)
write_metadata_block (wpc);
return TRUE;
}
// Note: The following function is no longer required because a proper wav
// header is now automatically generated for the application. However, if the
// application wants to generate its own header or wants to include additional
// chunks, then this function can still be used in which case the automatic
// wav header generation is suppressed.
// Add wrapper (currently RIFF only) to WavPack blocks. This should be called
// before sending any audio samples for the RIFF header or after all samples
// have been sent for any RIFF trailer. WavpackFlushSamples() should be called
// between sending the last samples and calling this for trailer data to make
// sure that headers and trailers don't get mixed up in very short files. If
// the exact contents of the RIFF header are not known because, for example,
// the file duration is uncertain or trailing chunks are possible, simply write
// a "dummy" header of the correct length. When all data has been written it
// will be possible to read the first block written and update the header
// directly. An example of this can be found in the Audition filter. A
// return of FALSE indicates an error.
int WavpackAddWrapper (WavpackContext *wpc, void *data, uint32_t bcount)
{
int64_t index = WavpackGetSampleIndex64 (wpc);
unsigned char meta_id;
if (!index || index == -1) {
wpc->riff_header_added = TRUE;
meta_id = wpc->file_format ? ID_ALT_HEADER : ID_RIFF_HEADER;
}
else {
wpc->riff_trailer_bytes += bcount;
meta_id = wpc->file_format ? ID_ALT_TRAILER : ID_RIFF_TRAILER;
}
return add_to_metadata (wpc, data, bcount, meta_id);
}
// Store computed MD5 sum in WavPack metadata. Note that the user must compute
// the 16 byte sum; it is not done here. A return of FALSE indicates an error.
// If any of the lower 8 bits of qmode are set, then this MD5 is stored with
// a metadata ID that old decoders do not recognize (because they would not
// interpret the qmode and would therefore fail the verification).
int WavpackStoreMD5Sum (WavpackContext *wpc, unsigned char data [16])
{
return add_to_metadata (wpc, data, 16, (wpc->config.qmode & 0xff) ? ID_ALT_MD5_CHECKSUM : ID_MD5_CHECKSUM);
}
#pragma pack(push,4)
typedef struct {
char ckID [4];
uint64_t chunkSize64;
} CS64Chunk;
typedef struct {
uint64_t riffSize64, dataSize64, sampleCount64;
uint32_t tableLength;
} DS64Chunk;
typedef struct {
char ckID [4];
uint32_t ckSize;
char junk [28];
} JunkChunk;
#pragma pack(pop)
#define DS64ChunkFormat "DDDL"
static int create_riff_header (WavpackContext *wpc, int64_t total_samples, void *outbuffer)
{
int do_rf64 = 0, write_junk = 1;
ChunkHeader ds64hdr, datahdr, fmthdr;
char *outptr = outbuffer;
RiffChunkHeader riffhdr;
DS64Chunk ds64_chunk;
JunkChunk junkchunk;
WaveHeader wavhdr;
int64_t total_data_bytes, total_riff_bytes;
int32_t channel_mask = wpc->config.channel_mask;
int32_t sample_rate = wpc->config.sample_rate;
int bytes_per_sample = wpc->config.bytes_per_sample;
int bits_per_sample = wpc->config.bits_per_sample;
int format = (wpc->config.float_norm_exp) ? 3 : 1;
int num_channels = wpc->config.num_channels;
int wavhdrsize = 16;
wpc->riff_header_created = TRUE;
if (format == 3 && wpc->config.float_norm_exp != 127) {
strcpy (wpc->error_message, "can't create valid RIFF wav header for non-normalized floating data!");
return FALSE;
}
if (total_samples == -1)
total_samples = 0x7ffff000 / (bytes_per_sample * num_channels);
total_data_bytes = total_samples * bytes_per_sample * num_channels;
if (total_data_bytes > 0xff000000) {
write_junk = 0;
do_rf64 = 1;
}
CLEAR (wavhdr);
wavhdr.FormatTag = format;
wavhdr.NumChannels = num_channels;
wavhdr.SampleRate = sample_rate;
wavhdr.BytesPerSecond = sample_rate * num_channels * bytes_per_sample;
wavhdr.BlockAlign = bytes_per_sample * num_channels;
wavhdr.BitsPerSample = bits_per_sample;
if (num_channels > 2 || channel_mask != 0x5 - num_channels) {
wavhdrsize = sizeof (wavhdr);
wavhdr.cbSize = 22;
wavhdr.ValidBitsPerSample = bits_per_sample;
wavhdr.SubFormat = format;
wavhdr.ChannelMask = channel_mask;
wavhdr.FormatTag = 0xfffe;
wavhdr.BitsPerSample = bytes_per_sample * 8;
wavhdr.GUID [4] = 0x10;
wavhdr.GUID [6] = 0x80;
wavhdr.GUID [9] = 0xaa;
wavhdr.GUID [11] = 0x38;
wavhdr.GUID [12] = 0x9b;
wavhdr.GUID [13] = 0x71;
}
memcpy (riffhdr.ckID, do_rf64 ? "RF64" : "RIFF", sizeof (riffhdr.ckID));
memcpy (riffhdr.formType, "WAVE", sizeof (riffhdr.formType));
total_riff_bytes = sizeof (riffhdr) + wavhdrsize + sizeof (datahdr) + total_data_bytes + wpc->riff_trailer_bytes;
if (do_rf64) total_riff_bytes += sizeof (ds64hdr) + sizeof (ds64_chunk);
if (write_junk) total_riff_bytes += sizeof (junkchunk);
memcpy (fmthdr.ckID, "fmt ", sizeof (fmthdr.ckID));
memcpy (datahdr.ckID, "data", sizeof (datahdr.ckID));
fmthdr.ckSize = wavhdrsize;
if (write_junk) {
CLEAR (junkchunk);
memcpy (junkchunk.ckID, "junk", sizeof (junkchunk.ckID));
junkchunk.ckSize = sizeof (junkchunk) - 8;
WavpackNativeToLittleEndian (&junkchunk, ChunkHeaderFormat);
}
if (do_rf64) {
memcpy (ds64hdr.ckID, "ds64", sizeof (ds64hdr.ckID));
ds64hdr.ckSize = sizeof (ds64_chunk);
CLEAR (ds64_chunk);
ds64_chunk.riffSize64 = total_riff_bytes;
ds64_chunk.dataSize64 = total_data_bytes;
ds64_chunk.sampleCount64 = total_samples;
riffhdr.ckSize = (uint32_t) -1;
datahdr.ckSize = (uint32_t) -1;
WavpackNativeToLittleEndian (&ds64hdr, ChunkHeaderFormat);
WavpackNativeToLittleEndian (&ds64_chunk, DS64ChunkFormat);
}
else {
riffhdr.ckSize = (uint32_t) total_riff_bytes;
datahdr.ckSize = (uint32_t) total_data_bytes;
}
WavpackNativeToLittleEndian (&riffhdr, ChunkHeaderFormat);
WavpackNativeToLittleEndian (&fmthdr, ChunkHeaderFormat);
WavpackNativeToLittleEndian (&wavhdr, WaveHeaderFormat);
WavpackNativeToLittleEndian (&datahdr, ChunkHeaderFormat);
// write the RIFF chunks up to just before the data starts
outptr = (char *) memcpy (outptr, &riffhdr, sizeof (riffhdr)) + sizeof (riffhdr);
if (do_rf64) {
outptr = (char *) memcpy (outptr, &ds64hdr, sizeof (ds64hdr)) + sizeof (ds64hdr);
outptr = (char *) memcpy (outptr, &ds64_chunk, sizeof (ds64_chunk)) + sizeof (ds64_chunk);
}
if (write_junk)
outptr = (char *) memcpy (outptr, &junkchunk, sizeof (junkchunk)) + sizeof (junkchunk);
outptr = (char *) memcpy (outptr, &fmthdr, sizeof (fmthdr)) + sizeof (fmthdr);
outptr = (char *) memcpy (outptr, &wavhdr, wavhdrsize) + wavhdrsize;
outptr = (char *) memcpy (outptr, &datahdr, sizeof (datahdr)) + sizeof (datahdr);
return (int)(outptr - (char *) outbuffer);
}
static int block_add_checksum (unsigned char *buffer_start, unsigned char *buffer_end, int bytes);
static int pack_streams (WavpackContext *wpc, uint32_t block_samples)
{
uint32_t max_blocksize, max_chans = 1, bcount;
unsigned char *outbuff, *outend, *out2buff, *out2end;
int result = TRUE, i;
// for calculating output (block) buffer size, first see if any streams are stereo
for (i = 0; i < wpc->num_streams; i++)
if (!(wpc->streams [i]->wphdr.flags & MONO_FLAG)) {
max_chans = 2;
break;
}
// then calculate maximum size based on bytes / sample
max_blocksize = block_samples * max_chans * ((wpc->streams [0]->wphdr.flags & BYTES_STORED) + 1);
// add margin based on how much "negative" compression is possible with pathological audio
if ((wpc->config.flags & CONFIG_FLOAT_DATA) && !(wpc->config.flags & CONFIG_SKIP_WVX))
max_blocksize += max_blocksize; // 100% margin for lossless float data
else
max_blocksize += max_blocksize >> 2; // otherwise 25% margin for everything else
max_blocksize += wpc->metabytes + 1024; // finally, add metadata & another 1K margin
max_blocksize += max_blocksize & 1; // and make sure it's even so we detect overflow
out2buff = (wpc->wvc_flag) ? malloc (max_blocksize) : NULL;
out2end = out2buff + max_blocksize;
outbuff = malloc (max_blocksize);
outend = outbuff + max_blocksize;
for (wpc->current_stream = 0; wpc->current_stream < wpc->num_streams; wpc->current_stream++) {
WavpackStream *wps = wpc->streams [wpc->current_stream];
uint32_t flags = wps->wphdr.flags;
flags &= ~MAG_MASK;
flags += (1 << MAG_LSB) * ((flags & BYTES_STORED) * 8 + 7);
SET_BLOCK_INDEX (wps->wphdr, wps->sample_index);
wps->wphdr.block_samples = block_samples;
wps->wphdr.flags = flags;
wps->block2buff = out2buff;
wps->block2end = out2end;
wps->blockbuff = outbuff;
wps->blockend = outend;
#ifdef ENABLE_DSD
if (flags & DSD_FLAG)
result = pack_dsd_block (wpc, wps->sample_buffer);
else
#endif
result = pack_block (wpc, wps->sample_buffer);
if (result) {
result = block_add_checksum (outbuff, outend, (flags & HYBRID_FLAG) ? 2 : 4);
if (result && out2buff)
result = block_add_checksum (out2buff, out2end, 2);
}
wps->blockbuff = wps->block2buff = NULL;
if (wps->wphdr.block_samples != block_samples)
block_samples = wps->wphdr.block_samples;
if (!result) {
strcpy (wpc->error_message, "output buffer overflowed!");
break;
}
bcount = ((WavpackHeader *) outbuff)->ckSize + 8;
WavpackNativeToLittleEndian ((WavpackHeader *) outbuff, WavpackHeaderFormat);
result = wpc->blockout (wpc->wv_out, outbuff, bcount);
if (!result) {
strcpy (wpc->error_message, "can't write WavPack data, disk probably full!");
break;
}
wpc->filelen += bcount;
if (out2buff) {
bcount = ((WavpackHeader *) out2buff)->ckSize + 8;
WavpackNativeToLittleEndian ((WavpackHeader *) out2buff, WavpackHeaderFormat);
result = wpc->blockout (wpc->wvc_out, out2buff, bcount);
if (!result) {
strcpy (wpc->error_message, "can't write WavPack data, disk probably full!");
break;
}
wpc->file2len += bcount;
}
if (wpc->acc_samples != block_samples)
memmove (wps->sample_buffer, wps->sample_buffer + block_samples * (flags & MONO_FLAG ? 1 : 2),
(wpc->acc_samples - block_samples) * sizeof (int32_t) * (flags & MONO_FLAG ? 1 : 2));
}
wpc->current_stream = 0;
wpc->ave_block_samples = (wpc->ave_block_samples * 0x7 + block_samples + 0x4) >> 3;
wpc->acc_samples -= block_samples;
free (outbuff);
if (out2buff)
free (out2buff);
return result;
}
// Given the pointer to the first block written (to either a .wv or .wvc file),
// update the block with the actual number of samples written. If the wav
// header was generated by the library, then it is updated also. This should
// be done if WavpackSetConfiguration() was called with an incorrect number
// of samples (or -1). It is the responsibility of the application to read and
// rewrite the block. An example of this can be found in the Audition filter.
static void block_update_checksum (unsigned char *buffer_start);
void WavpackUpdateNumSamples (WavpackContext *wpc, void *first_block)
{
uint32_t wrapper_size;
WavpackLittleEndianToNative (first_block, WavpackHeaderFormat);
SET_TOTAL_SAMPLES (* (WavpackHeader *) first_block, WavpackGetSampleIndex64 (wpc));
if (wpc->riff_header_created && WavpackGetWrapperLocation (first_block, &wrapper_size)) {
unsigned char riff_header [128];
if (wrapper_size == create_riff_header (wpc, WavpackGetSampleIndex64 (wpc), riff_header))
memcpy (WavpackGetWrapperLocation (first_block, NULL), riff_header, wrapper_size);
}
block_update_checksum (first_block);
WavpackNativeToLittleEndian (first_block, WavpackHeaderFormat);
}
// Note: The following function is no longer required because the wav header
// automatically generated for the application will also be updated by
// WavpackUpdateNumSamples (). However, if the application wants to generate
// its own header or wants to include additional chunks, then this function
// still must be used to update the application generated header.
// Given the pointer to the first block written to a WavPack file, this
// function returns the location of the stored RIFF header that was originally
// written with WavpackAddWrapper(). This would normally be used to update
// the wav header to indicate that a different number of samples was actually
// written or if additional RIFF chunks are written at the end of the file.
// The "size" parameter can be set to non-NULL to obtain the exact size of the
// RIFF header, and the function will return FALSE if the header is not found
// in the block's metadata (or it is not a valid WavPack block). It is the
// responsibility of the application to read and rewrite the block. An example
// of this can be found in the Audition filter.
static void *find_metadata (void *wavpack_block, int desired_id, uint32_t *size);
void *WavpackGetWrapperLocation (void *first_block, uint32_t *size)
{
void *loc;
WavpackLittleEndianToNative (first_block, WavpackHeaderFormat);
loc = find_metadata (first_block, ID_RIFF_HEADER, size);
if (!loc)
loc = find_metadata (first_block, ID_ALT_HEADER, size);
WavpackNativeToLittleEndian (first_block, WavpackHeaderFormat);
return loc;
}
static void *find_metadata (void *wavpack_block, int desired_id, uint32_t *size)
{
WavpackHeader *wphdr = wavpack_block;
unsigned char *dp, meta_id, c1, c2;
int32_t bcount, meta_bc;
if (strncmp (wphdr->ckID, "wvpk", 4))
return NULL;
bcount = wphdr->ckSize - sizeof (WavpackHeader) + 8;
dp = (unsigned char *)(wphdr + 1);
while (bcount >= 2) {
meta_id = *dp++;
c1 = *dp++;
meta_bc = c1 << 1;
bcount -= 2;
if (meta_id & ID_LARGE) {
if (bcount < 2)
break;
c1 = *dp++;
c2 = *dp++;
meta_bc += ((uint32_t) c1 << 9) + ((uint32_t) c2 << 17);
bcount -= 2;
}
if ((meta_id & ID_UNIQUE) == desired_id) {
if ((bcount - meta_bc) >= 0) {
if (size)
*size = meta_bc - ((meta_id & ID_ODD_SIZE) ? 1 : 0);
return dp;
}
else
return NULL;
}
bcount -= meta_bc;
dp += meta_bc;
}
return NULL;
}
int copy_metadata (WavpackMetadata *wpmd, unsigned char *buffer_start, unsigned char *buffer_end)
{
uint32_t mdsize = wpmd->byte_length + (wpmd->byte_length & 1);
WavpackHeader *wphdr = (WavpackHeader *) buffer_start;
mdsize += (wpmd->byte_length > 510) ? 4 : 2;
buffer_start += wphdr->ckSize + 8;
if (buffer_start + mdsize >= buffer_end)
return FALSE;
buffer_start [0] = wpmd->id | (wpmd->byte_length & 1 ? ID_ODD_SIZE : 0);
buffer_start [1] = (wpmd->byte_length + 1) >> 1;
if (wpmd->byte_length > 510) {
buffer_start [0] |= ID_LARGE;
buffer_start [2] = (wpmd->byte_length + 1) >> 9;
buffer_start [3] = (wpmd->byte_length + 1) >> 17;
}
if (wpmd->data && wpmd->byte_length) {
memcpy (buffer_start + (wpmd->byte_length > 510 ? 4 : 2), wpmd->data, wpmd->byte_length);
if (wpmd->byte_length & 1) // if size is odd, make sure pad byte is a zero
buffer_start [mdsize - 1] = 0;
}
wphdr->ckSize += mdsize;
return TRUE;
}
static int add_to_metadata (WavpackContext *wpc, void *data, uint32_t bcount, unsigned char id)
{
WavpackMetadata *mdp;
unsigned char *src = data;
while (bcount) {
if (wpc->metacount) {
uint32_t bc = bcount;
mdp = wpc->metadata + wpc->metacount - 1;
if (mdp->id == id) {
if (wpc->metabytes + bcount > 1000000)
bc = 1000000 - wpc->metabytes;
mdp->data = realloc (mdp->data, mdp->byte_length + bc);
memcpy ((char *) mdp->data + mdp->byte_length, src, bc);
mdp->byte_length += bc;
wpc->metabytes += bc;
bcount -= bc;
src += bc;
if (wpc->metabytes >= 1000000 && !write_metadata_block (wpc))
return FALSE;
}
}
if (bcount) {
wpc->metadata = realloc (wpc->metadata, (wpc->metacount + 1) * sizeof (WavpackMetadata));
mdp = wpc->metadata + wpc->metacount++;
mdp->byte_length = 0;
mdp->data = NULL;
mdp->id = id;
}
}
return TRUE;
}
static char *write_metadata (WavpackMetadata *wpmd, char *outdata)
{
unsigned char id = wpmd->id, wordlen [3];
wordlen [0] = (wpmd->byte_length + 1) >> 1;
wordlen [1] = (wpmd->byte_length + 1) >> 9;
wordlen [2] = (wpmd->byte_length + 1) >> 17;
if (wpmd->byte_length & 1)
id |= ID_ODD_SIZE;
if (wordlen [1] || wordlen [2])
id |= ID_LARGE;
*outdata++ = id;
*outdata++ = wordlen [0];
if (id & ID_LARGE) {
*outdata++ = wordlen [1];
*outdata++ = wordlen [2];
}
if (wpmd->data && wpmd->byte_length) {
memcpy (outdata, wpmd->data, wpmd->byte_length);
outdata += wpmd->byte_length;
if (wpmd->byte_length & 1)
*outdata++ = 0;
}
return outdata;
}
static int write_metadata_block (WavpackContext *wpc)
{
char *block_buff, *block_ptr;
WavpackHeader *wphdr;
if (wpc->metacount) {
int metacount = wpc->metacount, block_size = sizeof (WavpackHeader);
WavpackMetadata *wpmdp = wpc->metadata;
while (metacount--) {
block_size += wpmdp->byte_length + (wpmdp->byte_length & 1);
block_size += (wpmdp->byte_length > 510) ? 4 : 2;
wpmdp++;
}
// allocate 6 extra bytes for 4-byte checksum (which we add last)
wphdr = (WavpackHeader *) (block_buff = malloc (block_size + 6));
CLEAR (*wphdr);
memcpy (wphdr->ckID, "wvpk", 4);
SET_TOTAL_SAMPLES (*wphdr, wpc->total_samples);
wphdr->version = wpc->stream_version;
wphdr->ckSize = block_size - 8;
wphdr->block_samples = 0;
block_ptr = (char *)(wphdr + 1);
wpmdp = wpc->metadata;
while (wpc->metacount) {
block_ptr = write_metadata (wpmdp, block_ptr);
wpc->metabytes -= wpmdp->byte_length;
free_metadata (wpmdp++);
wpc->metacount--;
}
free (wpc->metadata);
wpc->metadata = NULL;
// add a 4-byte checksum here (increases block size by 6)
block_add_checksum ((unsigned char *) block_buff, (unsigned char *) block_buff + (block_size += 6), 4);
WavpackNativeToLittleEndian ((WavpackHeader *) block_buff, WavpackHeaderFormat);
if (!wpc->blockout (wpc->wv_out, block_buff, block_size)) {
free (block_buff);
strcpy (wpc->error_message, "can't write WavPack data, disk probably full!");
return FALSE;
}
free (block_buff);
}
return TRUE;
}
void free_metadata (WavpackMetadata *wpmd)
{
if (wpmd->data) {
free (wpmd->data);
wpmd->data = NULL;
}
}
// These two functions add or update the block checksums that were introduced in WavPack 5.0.
// The presence of the checksum is indicated by a flag in the wavpack header (HAS_CHECKSUM)
// and the actual metadata item should be the last one in the block, and can be either 2 or 4
// bytes. Of course, older versions of the decoder will simply ignore both of these.
static int block_add_checksum (unsigned char *buffer_start, unsigned char *buffer_end, int bytes)
{
WavpackHeader *wphdr = (WavpackHeader *) buffer_start;
#ifdef BITSTREAM_SHORTS
uint16_t *csptr = (uint16_t*) buffer_start;
#else
unsigned char *csptr = buffer_start;
#endif
int bcount = wphdr->ckSize + 8, wcount;
uint32_t csum = (uint32_t) -1;
if (bytes != 2 && bytes != 4)
return FALSE;
if (bcount < sizeof (WavpackHeader) || (bcount & 1) || buffer_start + bcount + 2 + bytes > buffer_end)
return FALSE;
wphdr->flags |= HAS_CHECKSUM;
wphdr->ckSize += 2 + bytes;
wcount = bcount >> 1;
#ifdef BITSTREAM_SHORTS
while (wcount--)
csum = (csum * 3) + *csptr++;
#else
WavpackNativeToLittleEndian ((WavpackHeader *) buffer_start, WavpackHeaderFormat);
while (wcount--) {
csum = (csum * 3) + csptr [0] + (csptr [1] << 8);
csptr += 2;
}
WavpackLittleEndianToNative ((WavpackHeader *) buffer_start, WavpackHeaderFormat);
#endif
buffer_start += bcount;
*buffer_start++ = ID_BLOCK_CHECKSUM;
*buffer_start++ = bytes >> 1;
if (bytes == 4) {
*buffer_start++ = csum;
*buffer_start++ = csum >> 8;
*buffer_start++ = csum >> 16;
*buffer_start++ = csum >> 24;
}
else {
csum ^= csum >> 16;
*buffer_start++ = csum;
*buffer_start++ = csum >> 8;
}
return TRUE;
}
static void block_update_checksum (unsigned char *buffer_start)
{
WavpackHeader *wphdr = (WavpackHeader *) buffer_start;
unsigned char *dp, meta_id, c1, c2;
uint32_t bcount, meta_bc;
if (!(wphdr->flags & HAS_CHECKSUM))
return;
bcount = wphdr->ckSize - sizeof (WavpackHeader) + 8;
dp = (unsigned char *)(wphdr + 1);
while (bcount >= 2) {
meta_id = *dp++;
c1 = *dp++;
meta_bc = c1 << 1;
bcount -= 2;
if (meta_id & ID_LARGE) {
if (bcount < 2)
return;
c1 = *dp++;
c2 = *dp++;
meta_bc += ((uint32_t) c1 << 9) + ((uint32_t) c2 << 17);
bcount -= 2;
}
if (bcount < meta_bc)
return;
if ((meta_id & ID_UNIQUE) == ID_BLOCK_CHECKSUM) {
#ifdef BITSTREAM_SHORTS
uint16_t *csptr = (uint16_t*) buffer_start;
#else
unsigned char *csptr = buffer_start;
#endif
int wcount = (int)(dp - 2 - buffer_start) >> 1;
uint32_t csum = (uint32_t) -1;
if ((meta_id & ID_ODD_SIZE) || meta_bc < 2 || meta_bc > 4)
return;
#ifdef BITSTREAM_SHORTS
while (wcount--)
csum = (csum * 3) + *csptr++;
#else
WavpackNativeToLittleEndian ((WavpackHeader *) buffer_start, WavpackHeaderFormat);
while (wcount--) {
csum = (csum * 3) + csptr [0] + (csptr [1] << 8);
csptr += 2;
}
WavpackLittleEndianToNative ((WavpackHeader *) buffer_start, WavpackHeaderFormat);
#endif
if (meta_bc == 4) {
*dp++ = csum;
*dp++ = csum >> 8;
*dp++ = csum >> 16;
*dp++ = csum >> 24;
return;
}
else {
csum ^= csum >> 16;
*dp++ = csum;
*dp++ = csum >> 8;
return;
}
}
bcount -= meta_bc;
dp += meta_bc;
}
}