diff options
| author | ayumi <ayumi@noreply.codeberg.org> | 2025-01-31 19:08:39 +0100 |
|---|---|---|
| committer | ayumi <ayumi@noreply.codeberg.org> | 2025-03-13 03:29:03 +0100 |
| commit | 885eb1812c15263ad759741ad138cf7188fdf739 (patch) | |
| tree | 24aff12a5d67f77675281fd70c0857164e913331 /lib | |
| parent | a3639860761dcdb5ef9c31bb34497f32cadd9ff3 (diff) | |
| download | tangara-fw-885eb1812c15263ad759741ad138cf7188fdf739.tar.gz | |
Add WavPack support
Diffstat (limited to 'lib')
| -rw-r--r-- | lib/wavpack/CMakeLists.txt | 4 | ||||
| -rw-r--r-- | lib/wavpack/bits.c | 141 | ||||
| -rw-r--r-- | lib/wavpack/float.c | 50 | ||||
| -rw-r--r-- | lib/wavpack/license.txt | 25 | ||||
| -rw-r--r-- | lib/wavpack/metadata.c | 105 | ||||
| -rw-r--r-- | lib/wavpack/readme.txt | 68 | ||||
| -rw-r--r-- | lib/wavpack/unpack.c | 785 | ||||
| -rw-r--r-- | lib/wavpack/wavpack.h | 394 | ||||
| -rw-r--r-- | lib/wavpack/words.c | 560 | ||||
| -rw-r--r-- | lib/wavpack/wputils.c | 350 | ||||
| -rw-r--r-- | lib/wavpack/wvfilter.c | 200 |
11 files changed, 2682 insertions, 0 deletions
diff --git a/lib/wavpack/CMakeLists.txt b/lib/wavpack/CMakeLists.txt new file mode 100644 index 00000000..98fcda95 --- /dev/null +++ b/lib/wavpack/CMakeLists.txt @@ -0,0 +1,4 @@ +idf_component_register( + SRCS bits.c float.c wputils.c metadata.c unpack.c words.c + INCLUDE_DIRS . +) diff --git a/lib/wavpack/bits.c b/lib/wavpack/bits.c new file mode 100644 index 00000000..d69cd288 --- /dev/null +++ b/lib/wavpack/bits.c @@ -0,0 +1,141 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// bits.c
+
+// This module provides utilities to support the BitStream structure which is
+// used to read and write all WavPack audio data streams. It also contains a
+// wrapper for the stream I/O functions and a set of functions dealing with
+// endian-ness, both for enhancing portability. Finally, a debug wrapper for
+// the malloc() system is provided.
+
+#include "wavpack.h"
+
+#include <string.h>
+#include <ctype.h>
+
+////////////////////////// Bitstream functions ////////////////////////////////
+
+// Open the specified BitStream and associate with the specified buffer.
+
+static void bs_read (Bitstream *bs);
+
+void bs_open_read (Bitstream *bs, uchar *buffer_start, uchar *buffer_end, read_stream file, void *user_data, uint32_t file_bytes)
+{
+ CLEAR (*bs);
+ bs->buf = buffer_start;
+ bs->end = buffer_end;
+
+ if (file) {
+ bs->ptr = bs->end - 1;
+ bs->file_bytes = file_bytes;
+ bs->file = file;
+ bs->user_data = user_data;
+ }
+ else
+ bs->ptr = bs->buf - 1;
+
+ bs->wrap = bs_read;
+}
+
+// This function is only called from the getbit() and getbits() macros when
+// the BitStream has been exhausted and more data is required. Sinve these
+// bistreams no longer access files, this function simple sets an error and
+// resets the buffer.
+
+static void bs_read (Bitstream *bs)
+{
+ if (bs->file && bs->file_bytes) {
+ uint32_t bytes_read, bytes_to_read = bs->end - bs->buf;
+
+ if (bytes_to_read > bs->file_bytes)
+ bytes_to_read = bs->file_bytes;
+
+ bytes_read = bs->file (bs->user_data, bs->buf, bytes_to_read);
+
+ if (bytes_read) {
+ bs->end = bs->buf + bytes_read;
+ bs->file_bytes -= bytes_read;
+ }
+ else {
+ memset (bs->buf, -1, bs->end - bs->buf);
+ bs->error = 1;
+ }
+ }
+ else
+ bs->error = 1;
+
+ if (bs->error)
+ memset (bs->buf, -1, bs->end - bs->buf);
+
+ bs->ptr = bs->buf;
+}
+
+/////////////////////// Endian Correction Routines ////////////////////////////
+
+void little_endian_to_native (void *data, char *format)
+{
+ uchar *cp = (uchar *) data;
+ int32_t temp;
+
+ while (*format) {
+ switch (*format) {
+ case 'L':
+ temp = cp [0] + ((int32_t) cp [1] << 8) + ((int32_t) cp [2] << 16) + ((int32_t) cp [3] << 24);
+ * (int32_t *) cp = temp;
+ cp += 4;
+ break;
+
+ case 'S':
+ temp = cp [0] + (cp [1] << 8);
+ * (short *) cp = (short) temp;
+ cp += 2;
+ break;
+
+ default:
+ if (isdigit ((unsigned char) *format))
+ cp += *format - '0';
+
+ break;
+ }
+
+ format++;
+ }
+}
+
+void native_to_little_endian (void *data, char *format)
+{
+ uchar *cp = (uchar *) data;
+ int32_t temp;
+
+ while (*format) {
+ switch (*format) {
+ case 'L':
+ temp = * (int32_t *) cp;
+ *cp++ = (uchar) temp;
+ *cp++ = (uchar) (temp >> 8);
+ *cp++ = (uchar) (temp >> 16);
+ *cp++ = (uchar) (temp >> 24);
+ break;
+
+ case 'S':
+ temp = * (short *) cp;
+ *cp++ = (uchar) temp;
+ *cp++ = (uchar) (temp >> 8);
+ break;
+
+ default:
+ if (isdigit ((unsigned char) *format))
+ cp += *format - '0';
+
+ break;
+ }
+
+ format++;
+ }
+}
diff --git a/lib/wavpack/float.c b/lib/wavpack/float.c new file mode 100644 index 00000000..4b9b44ee --- /dev/null +++ b/lib/wavpack/float.c @@ -0,0 +1,50 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// float.c
+
+#include "wavpack.h"
+
+int read_float_info (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+ int bytecnt = wpmd->byte_length;
+ char *byteptr = wpmd->data;
+
+ if (bytecnt != 4)
+ return FALSE;
+
+ wps->float_flags = *byteptr++;
+ wps->float_shift = *byteptr++;
+ wps->float_max_exp = *byteptr++;
+ wps->float_norm_exp = *byteptr;
+ return TRUE;
+}
+
+void float_values (WavpackStream *wps, int32_t *values, int32_t num_values)
+{
+ int shift = wps->float_max_exp - wps->float_norm_exp + wps->float_shift;
+
+ if (shift > 32)
+ shift = 32;
+ else if (shift < -32)
+ shift = -32;
+
+ while (num_values--) {
+ if (shift > 0)
+ *values <<= shift;
+ else if (shift < 0)
+ *values >>= -shift;
+
+ if (*values > 8388607L)
+ *values = 8388607L;
+ else if (*values < -8388608L)
+ *values = -8388608L;
+
+ values++;
+ }
+}
diff --git a/lib/wavpack/license.txt b/lib/wavpack/license.txt new file mode 100644 index 00000000..98f6e6b1 --- /dev/null +++ b/lib/wavpack/license.txt @@ -0,0 +1,25 @@ + Copyright (c) 1998 - 2006 Conifer Software
+ All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are met:
+
+ * Redistributions of source code must retain the above copyright notice,
+ this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ this list of conditions and the following disclaimer in the
+ documentation and/or other materials provided with the distribution.
+ * Neither the name of Conifer Software nor the names of its contributors
+ may be used to endorse or promote products derived from this software
+ without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE FOR
+ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/lib/wavpack/metadata.c b/lib/wavpack/metadata.c new file mode 100644 index 00000000..b80e905a --- /dev/null +++ b/lib/wavpack/metadata.c @@ -0,0 +1,105 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// metadata.c
+
+// This module handles the metadata structure introduced in WavPack 4.0
+
+#include "wavpack.h"
+
+int read_metadata_buff (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+ uchar tchar;
+
+ if (!wpc->infile (wpc->user_data, &wpmd->id, 1) || !wpc->infile (wpc->user_data, &tchar, 1))
+ return FALSE;
+
+ wpmd->byte_length = tchar << 1;
+
+ if (wpmd->id & ID_LARGE) {
+ wpmd->id &= ~ID_LARGE;
+
+ if (!wpc->infile (wpc->user_data, &tchar, 1))
+ return FALSE;
+
+ wpmd->byte_length += (int32_t) tchar << 9;
+
+ if (!wpc->infile (wpc->user_data, &tchar, 1))
+ return FALSE;
+
+ wpmd->byte_length += (int32_t) tchar << 17;
+ }
+
+ if (wpmd->id & ID_ODD_SIZE) {
+ wpmd->id &= ~ID_ODD_SIZE;
+ wpmd->byte_length--;
+ }
+
+ if (wpmd->byte_length && wpmd->byte_length <= sizeof (wpc->read_buffer)) {
+ uint32_t bytes_to_read = wpmd->byte_length + (wpmd->byte_length & 1);
+
+ if (wpc->infile (wpc->user_data, wpc->read_buffer, bytes_to_read) != (int32_t) bytes_to_read) {
+ wpmd->data = NULL;
+ return FALSE;
+ }
+
+ wpmd->data = wpc->read_buffer;
+ }
+ else
+ wpmd->data = NULL;
+
+ return TRUE;
+}
+
+int process_metadata (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+ WavpackStream *wps = &wpc->stream;
+
+ switch (wpmd->id) {
+ case ID_DUMMY:
+ return TRUE;
+
+ case ID_DECORR_TERMS:
+ return read_decorr_terms (wps, wpmd);
+
+ case ID_DECORR_WEIGHTS:
+ return read_decorr_weights (wps, wpmd);
+
+ case ID_DECORR_SAMPLES:
+ return read_decorr_samples (wps, wpmd);
+
+ case ID_ENTROPY_VARS:
+ return read_entropy_vars (wps, wpmd);
+
+ case ID_HYBRID_PROFILE:
+ return read_hybrid_profile (wps, wpmd);
+
+ case ID_FLOAT_INFO:
+ return read_float_info (wps, wpmd);
+
+ case ID_INT32_INFO:
+ return read_int32_info (wps, wpmd);
+
+ case ID_CHANNEL_INFO:
+ return read_channel_info (wpc, wpmd);
+
+ case ID_CONFIG_BLOCK:
+ return read_config_info (wpc, wpmd);
+
+ case ID_WV_BITSTREAM:
+ return init_wv_bitstream (wpc, wpmd);
+
+ case ID_SHAPING_WEIGHTS:
+ case ID_WVC_BITSTREAM:
+ case ID_WVX_BITSTREAM:
+ return TRUE;
+
+ default:
+ return (wpmd->id & ID_OPTIONAL_DATA) ? TRUE : FALSE;
+ }
+}
diff --git a/lib/wavpack/readme.txt b/lib/wavpack/readme.txt new file mode 100644 index 00000000..4ccbdf42 --- /dev/null +++ b/lib/wavpack/readme.txt @@ -0,0 +1,68 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+This package contains a tiny version of the WavPack 4.40 decoder that might
+be used in a "resource limited" CPU environment or form the basis for a
+hardware decoding implementation. It is packaged with a demo command-line
+program that accepts a WavPack audio file on stdin and outputs a RIFF wav
+file to stdout. The program is standard C, and a win32 executable is
+included which was compiled under MS Visual C++ 6.0 using this command:
+
+cl /O1 /DWIN32 wvfilter.c wputils.c unpack.c float.c metadata.c words.c bits.c
+
+WavPack data is read with a stream reading callback. No direct seeking is
+provided for, but it is possible to start decoding anywhere in a WavPack
+stream. In this case, WavPack will be able to provide the sample-accurate
+position when it synchs with the data and begins decoding. The WIN32 macro
+is used for Windows to force the stdin and stdout streams to be binary mode.
+
+Compared to the previous version, this library has been optimized somewhat
+for improved performance in exchange for slightly larger code size. The
+library also now includes hand-optimized assembly language versions of the
+decorrelation functions for both the ColdFire (w/EMAC) and ARM processors.
+
+For demonstration purposes this uses a single static copy of the
+WavpackContext structure, so obviously it cannot be used for more than one
+file at a time. Also, this decoder will not handle "correction" files, plays
+only the first two channels of multi-channel files, and is limited in
+resolution in some large integer or floating point files (but always
+provides at least 24 bits of resolution). It also will not accept WavPack
+files from before version 4.0.
+
+The previous version of this library would handle float files by returning
+32-bit floating-point data (even though no floating point math was used).
+Because this library would normally be used for simply playing WavPack
+files where lossless performance (beyond 24-bits) is not relevant, I have
+changed this behavior. Now, these files will generate clipped 24-bit data.
+The MODE_FLOAT flag will still be returned by WavpackGetMode(), but the
+BitsPerSample and BytesPerSample queries will be 24 and 3, respectfully.
+What this means is that an application that can handle 24-bit data will
+now be able to handle floating point data (assuming that the MODE_FLOAT
+flag is ignored).
+
+To make this code viable on the greatest number of hardware platforms, the
+following are true:
+
+ speed is about 5x realtime on an AMD K6 300 MHz
+ ("high" mode 16/44 stereo; normal mode is about twice that fast)
+
+ no floating-point math required; just 32b * 32b = 32b int multiply
+
+ large data areas are static and less than 4K total
+ executable code and tables are less than 40K
+ no malloc / free usage
+
+To maintain compatibility on various platforms, the following conventions
+are used:
+
+ a "char" must be exactly 8-bits
+ a "short" must be exactly 16-bits
+ an "int" must be at least 16-bits, but may be larger
+ the "long" type is not used to avoid problems with 64-bit compilers
+
+Questions or comments should be directed to david@wavpack.com
diff --git a/lib/wavpack/unpack.c b/lib/wavpack/unpack.c new file mode 100644 index 00000000..e169c47f --- /dev/null +++ b/lib/wavpack/unpack.c @@ -0,0 +1,785 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// unpack.c
+
+// This module actually handles the decompression of the audio data, except
+// for the entropy decoding which is handled by the words.c module. For
+// maximum efficiency, the conversion is isolated to tight loops that handle
+// an entire buffer.
+
+#include "wavpack.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#define LOSSY_MUTE
+
+///////////////////////////// executable code ////////////////////////////////
+
+// This function initializes everything required to unpack a WavPack block
+// and must be called before unpack_samples() is called to obtain audio data.
+// It is assumed that the WavpackHeader has been read into the wps->wphdr
+// (in the current WavpackStream). This is where all the metadata blocks are
+// scanned up to the one containing the audio bitstream.
+
+int unpack_init (WavpackContext *wpc)
+{
+ WavpackStream *wps = &wpc->stream;
+ WavpackMetadata wpmd;
+
+ if (wps->wphdr.block_samples && wps->wphdr.block_index != (uint32_t) -1)
+ wps->sample_index = wps->wphdr.block_index;
+
+ wps->mute_error = FALSE;
+ wps->crc = 0xffffffff;
+ CLEAR (wps->wvbits);
+ CLEAR (wps->decorr_passes);
+ CLEAR (wps->w);
+
+ while (read_metadata_buff (wpc, &wpmd)) {
+ if (!process_metadata (wpc, &wpmd)) {
+ strcpy (wpc->error_message, "invalid metadata!");
+ return FALSE;
+ }
+
+ if (wpmd.id == ID_WV_BITSTREAM)
+ break;
+ }
+
+ if (wps->wphdr.block_samples && !bs_is_open (&wps->wvbits)) {
+ strcpy (wpc->error_message, "invalid WavPack file!");
+ return FALSE;
+ }
+
+ if (wps->wphdr.block_samples) {
+ if ((wps->wphdr.flags & INT32_DATA) && wps->int32_sent_bits)
+ wpc->lossy_blocks = TRUE;
+
+ if ((wps->wphdr.flags & FLOAT_DATA) &&
+ wps->float_flags & (FLOAT_EXCEPTIONS | FLOAT_ZEROS_SENT | FLOAT_SHIFT_SENT | FLOAT_SHIFT_SAME))
+ wpc->lossy_blocks = TRUE;
+ }
+
+ return TRUE;
+}
+
+// This function initialzes the main bitstream for audio samples, which must
+// be in the "wv" file.
+
+int init_wv_bitstream (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+ WavpackStream *wps = &wpc->stream;
+
+ if (wpmd->data)
+ bs_open_read (&wps->wvbits, wpmd->data, (unsigned char *) wpmd->data + wpmd->byte_length, NULL, NULL, 0);
+ else if (wpmd->byte_length)
+ bs_open_read (&wps->wvbits, wpc->read_buffer, wpc->read_buffer + sizeof (wpc->read_buffer),
+ wpc->infile, wpc->user_data, wpmd->byte_length + (wpmd->byte_length & 1));
+
+ return TRUE;
+}
+
+// Read decorrelation terms from specified metadata block into the
+// decorr_passes array. The terms range from -3 to 8, plus 17 & 18;
+// other values are reserved and generate errors for now. The delta
+// ranges from 0 to 7 with all values valid. Note that the terms are
+// stored in the opposite order in the decorr_passes array compared
+// to packing.
+
+int read_decorr_terms (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+ int termcnt = wpmd->byte_length;
+ uchar *byteptr = wpmd->data;
+ struct decorr_pass *dpp;
+
+ if (termcnt > MAX_NTERMS)
+ return FALSE;
+
+ wps->num_terms = termcnt;
+
+ for (dpp = wps->decorr_passes + termcnt - 1; termcnt--; dpp--) {
+ dpp->term = (int)(*byteptr & 0x1f) - 5;
+ dpp->delta = (*byteptr++ >> 5) & 0x7;
+
+ if (!dpp->term || dpp->term < -3 || (dpp->term > MAX_TERM && dpp->term < 17) || dpp->term > 18)
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+// Read decorrelation weights from specified metadata block into the
+// decorr_passes array. The weights range +/-1024, but are rounded and
+// truncated to fit in signed chars for metadata storage. Weights are
+// separate for the two channels and are specified from the "last" term
+// (first during encode). Unspecified weights are set to zero.
+
+int read_decorr_weights (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+ int termcnt = wpmd->byte_length, tcount;
+ signed char *byteptr = wpmd->data;
+ struct decorr_pass *dpp;
+
+ if (!(wps->wphdr.flags & MONO_DATA))
+ termcnt /= 2;
+
+ if (termcnt > wps->num_terms)
+ return FALSE;
+
+ for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
+ dpp->weight_A = dpp->weight_B = 0;
+
+ while (--dpp >= wps->decorr_passes && termcnt--) {
+ dpp->weight_A = restore_weight (*byteptr++);
+
+ if (!(wps->wphdr.flags & MONO_DATA))
+ dpp->weight_B = restore_weight (*byteptr++);
+ }
+
+ return TRUE;
+}
+
+// Read decorrelation samples from specified metadata block into the
+// decorr_passes array. The samples are signed 32-bit values, but are
+// converted to signed log2 values for storage in metadata. Values are
+// stored for both channels and are specified from the "last" term
+// (first during encode) with unspecified samples set to zero. The
+// number of samples stored varies with the actual term value, so
+// those must obviously come first in the metadata.
+
+int read_decorr_samples (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+ uchar *byteptr = wpmd->data;
+ uchar *endptr = byteptr + wpmd->byte_length;
+ struct decorr_pass *dpp;
+ int tcount;
+
+ for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
+ CLEAR (dpp->samples_A);
+ CLEAR (dpp->samples_B);
+ }
+
+ if (wps->wphdr.version == 0x402 && (wps->wphdr.flags & HYBRID_FLAG)) {
+ byteptr += 2;
+
+ if (!(wps->wphdr.flags & MONO_DATA))
+ byteptr += 2;
+ }
+
+ while (dpp-- > wps->decorr_passes && byteptr < endptr)
+ if (dpp->term > MAX_TERM) {
+ dpp->samples_A [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+ dpp->samples_A [1] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
+ byteptr += 4;
+
+ if (!(wps->wphdr.flags & MONO_DATA)) {
+ dpp->samples_B [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+ dpp->samples_B [1] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
+ byteptr += 4;
+ }
+ }
+ else if (dpp->term < 0) {
+ dpp->samples_A [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+ dpp->samples_B [0] = exp2s ((short)(byteptr [2] + (byteptr [3] << 8)));
+ byteptr += 4;
+ }
+ else {
+ int m = 0, cnt = dpp->term;
+
+ while (cnt--) {
+ dpp->samples_A [m] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+ byteptr += 2;
+
+ if (!(wps->wphdr.flags & MONO_DATA)) {
+ dpp->samples_B [m] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+ byteptr += 2;
+ }
+
+ m++;
+ }
+ }
+
+ return byteptr == endptr;
+}
+
+// Read the int32 data from the specified metadata into the specified stream.
+// This data is used for integer data that has more than 24 bits of magnitude
+// or, in some cases, used to eliminate redundant bits from any audio stream.
+
+int read_int32_info (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+ int bytecnt = wpmd->byte_length;
+ char *byteptr = wpmd->data;
+
+ if (bytecnt != 4)
+ return FALSE;
+
+ wps->int32_sent_bits = *byteptr++;
+ wps->int32_zeros = *byteptr++;
+ wps->int32_ones = *byteptr++;
+ wps->int32_dups = *byteptr;
+ return TRUE;
+}
+
+// Read multichannel information from metadata. The first byte is the total
+// number of channels and the following bytes represent the channel_mask
+// as described for Microsoft WAVEFORMATEX.
+
+int read_channel_info (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+ int bytecnt = wpmd->byte_length, shift = 0;
+ char *byteptr = wpmd->data;
+ uint32_t mask = 0;
+
+ if (!bytecnt || bytecnt > 5)
+ return FALSE;
+
+ wpc->config.num_channels = *byteptr++;
+
+ while (--bytecnt) {
+ mask |= (uint32_t) *byteptr++ << shift;
+ shift += 8;
+ }
+
+ wpc->config.channel_mask = mask;
+ return TRUE;
+}
+
+// Read configuration information from metadata.
+
+int read_config_info (WavpackContext *wpc, WavpackMetadata *wpmd)
+{
+ int bytecnt = wpmd->byte_length;
+ uchar *byteptr = wpmd->data;
+
+ if (bytecnt >= 3) {
+ wpc->config.flags &= 0xff;
+ wpc->config.flags |= (int32_t) *byteptr++ << 8;
+ wpc->config.flags |= (int32_t) *byteptr++ << 16;
+ wpc->config.flags |= (int32_t) *byteptr << 24;
+ }
+
+ return TRUE;
+}
+
+// This monster actually unpacks the WavPack bitstream(s) into the specified
+// buffer as 32-bit integers or floats (depending on orignal data). Lossy
+// samples will be clipped to their original limits (i.e. 8-bit samples are
+// clipped to -128/+127) but are still returned in int32_ts. It is up to the
+// caller to potentially reformat this for the final output including any
+// multichannel distribution, block alignment or endian compensation. The
+// function unpack_init() must have been called and the entire WavPack block
+// must still be visible (although wps->blockbuff will not be accessed again).
+// For maximum clarity, the function is broken up into segments that handle
+// various modes. This makes for a few extra infrequent flag checks, but
+// makes the code easier to follow because the nesting does not become so
+// deep. For maximum efficiency, the conversion is isolated to tight loops
+// that handle an entire buffer. The function returns the total number of
+// samples unpacked, which can be less than the number requested if an error
+// occurs or the end of the block is reached.
+
+#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
+extern void decorr_stereo_pass_cont_mcf5249 (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
+#elif defined(CPU_ARM) && !defined(SIMULATOR)
+extern void decorr_stereo_pass_cont_arm (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
+extern void decorr_stereo_pass_cont_arml (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
+#else
+static void decorr_stereo_pass_cont (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
+#endif
+
+static void decorr_mono_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
+static void decorr_stereo_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count);
+static void fixup_samples (WavpackStream *wps, int32_t *buffer, uint32_t sample_count);
+
+int32_t unpack_samples (WavpackContext *wpc, int32_t *buffer, uint32_t sample_count)
+{
+ WavpackStream *wps = &wpc->stream;
+ uint32_t flags = wps->wphdr.flags, crc = wps->crc, i;
+ int32_t mute_limit = (1L << ((flags & MAG_MASK) >> MAG_LSB)) + 2;
+ struct decorr_pass *dpp;
+ int32_t *bptr, *eptr;
+ int tcount;
+
+ if (wps->sample_index + sample_count > wps->wphdr.block_index + wps->wphdr.block_samples)
+ sample_count = wps->wphdr.block_index + wps->wphdr.block_samples - wps->sample_index;
+
+ if (wps->mute_error) {
+ memset (buffer, 0, sample_count * (flags & MONO_FLAG ? 4 : 8));
+ wps->sample_index += sample_count;
+ return sample_count;
+ }
+
+ if (flags & HYBRID_FLAG)
+ mute_limit *= 2;
+
+ ///////////////////// handle version 4 mono data /////////////////////////
+
+ if (flags & MONO_DATA) {
+ eptr = buffer + sample_count;
+ i = get_words (buffer, sample_count, flags, &wps->w, &wps->wvbits);
+
+ for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
+ decorr_mono_pass (dpp, buffer, sample_count);
+
+ for (bptr = buffer; bptr < eptr; ++bptr) {
+ if (labs (bptr [0]) > mute_limit) {
+ i = bptr - buffer;
+ break;
+ }
+
+ crc = crc * 3 + bptr [0];
+ }
+ }
+
+ //////////////////// handle version 4 stereo data ////////////////////////
+
+ else {
+ eptr = buffer + (sample_count * 2);
+ i = get_words (buffer, sample_count, flags, &wps->w, &wps->wvbits);
+
+ if (sample_count < 16)
+ for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++)
+ decorr_stereo_pass (dpp, buffer, sample_count);
+ else
+ for (tcount = wps->num_terms, dpp = wps->decorr_passes; tcount--; dpp++) {
+ decorr_stereo_pass (dpp, buffer, 8);
+#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
+ decorr_stereo_pass_cont_mcf5249 (dpp, buffer + 16, sample_count - 8);
+#elif defined(CPU_ARM) && !defined(SIMULATOR)
+ if (((flags & MAG_MASK) >> MAG_LSB) > 15)
+ decorr_stereo_pass_cont_arml (dpp, buffer + 16, sample_count - 8);
+ else
+ decorr_stereo_pass_cont_arm (dpp, buffer + 16, sample_count - 8);
+#else
+ decorr_stereo_pass_cont (dpp, buffer + 16, sample_count - 8);
+#endif
+ }
+
+ if (flags & JOINT_STEREO)
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ bptr [0] += (bptr [1] -= (bptr [0] >> 1));
+
+ if (labs (bptr [0]) > mute_limit || labs (bptr [1]) > mute_limit) {
+ i = (bptr - buffer) / 2;
+ break;
+ }
+
+ crc = (crc * 3 + bptr [0]) * 3 + bptr [1];
+ }
+ else
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ if (labs (bptr [0]) > mute_limit || labs (bptr [1]) > mute_limit) {
+ i = (bptr - buffer) / 2;
+ break;
+ }
+
+ crc = (crc * 3 + bptr [0]) * 3 + bptr [1];
+ }
+ }
+
+ if (i != sample_count) {
+ memset (buffer, 0, sample_count * (flags & MONO_FLAG ? 4 : 8));
+ wps->mute_error = TRUE;
+ i = sample_count;
+ }
+
+ fixup_samples (wps, buffer, i);
+
+ if (flags & FALSE_STEREO) {
+ int32_t *dptr = buffer + i * 2;
+ int32_t *sptr = buffer + i;
+ int32_t c = i;
+
+ while (c--) {
+ *--dptr = *--sptr;
+ *--dptr = *sptr;
+ }
+ }
+
+ wps->sample_index += i;
+ wps->crc = crc;
+
+ return i;
+}
+
+static void decorr_stereo_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
+{
+ int32_t delta = dpp->delta, weight_A = dpp->weight_A, weight_B = dpp->weight_B;
+ int32_t *bptr, *eptr = buffer + (sample_count * 2), sam_A, sam_B;
+ int m, k;
+
+ switch (dpp->term) {
+
+ case 17:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_A = 2 * dpp->samples_A [0] - dpp->samples_A [1];
+ dpp->samples_A [1] = dpp->samples_A [0];
+ dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
+ update_weight (weight_A, delta, sam_A, bptr [0]);
+ bptr [0] = dpp->samples_A [0];
+
+ sam_A = 2 * dpp->samples_B [0] - dpp->samples_B [1];
+ dpp->samples_B [1] = dpp->samples_B [0];
+ dpp->samples_B [0] = apply_weight (weight_B, sam_A) + bptr [1];
+ update_weight (weight_B, delta, sam_A, bptr [1]);
+ bptr [1] = dpp->samples_B [0];
+ }
+
+ break;
+
+ case 18:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_A = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
+ dpp->samples_A [1] = dpp->samples_A [0];
+ dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
+ update_weight (weight_A, delta, sam_A, bptr [0]);
+ bptr [0] = dpp->samples_A [0];
+
+ sam_A = (3 * dpp->samples_B [0] - dpp->samples_B [1]) >> 1;
+ dpp->samples_B [1] = dpp->samples_B [0];
+ dpp->samples_B [0] = apply_weight (weight_B, sam_A) + bptr [1];
+ update_weight (weight_B, delta, sam_A, bptr [1]);
+ bptr [1] = dpp->samples_B [0];
+ }
+
+ break;
+
+ default:
+ for (m = 0, k = dpp->term & (MAX_TERM - 1), bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_A = dpp->samples_A [m];
+ dpp->samples_A [k] = apply_weight (weight_A, sam_A) + bptr [0];
+ update_weight (weight_A, delta, sam_A, bptr [0]);
+ bptr [0] = dpp->samples_A [k];
+
+ sam_A = dpp->samples_B [m];
+ dpp->samples_B [k] = apply_weight (weight_B, sam_A) + bptr [1];
+ update_weight (weight_B, delta, sam_A, bptr [1]);
+ bptr [1] = dpp->samples_B [k];
+
+ m = (m + 1) & (MAX_TERM - 1);
+ k = (k + 1) & (MAX_TERM - 1);
+ }
+
+ if (m) {
+ int32_t temp_samples [MAX_TERM];
+
+ memcpy (temp_samples, dpp->samples_A, sizeof (dpp->samples_A));
+
+ for (k = 0; k < MAX_TERM; k++, m++)
+ dpp->samples_A [k] = temp_samples [m & (MAX_TERM - 1)];
+
+ memcpy (temp_samples, dpp->samples_B, sizeof (dpp->samples_B));
+
+ for (k = 0; k < MAX_TERM; k++, m++)
+ dpp->samples_B [k] = temp_samples [m & (MAX_TERM - 1)];
+ }
+
+ break;
+
+ case -1:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_A = bptr [0] + apply_weight (weight_A, dpp->samples_A [0]);
+ update_weight_clip (weight_A, delta, dpp->samples_A [0], bptr [0]);
+ bptr [0] = sam_A;
+ dpp->samples_A [0] = bptr [1] + apply_weight (weight_B, sam_A);
+ update_weight_clip (weight_B, delta, sam_A, bptr [1]);
+ bptr [1] = dpp->samples_A [0];
+ }
+
+ break;
+
+ case -2:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_B = bptr [1] + apply_weight (weight_B, dpp->samples_B [0]);
+ update_weight_clip (weight_B, delta, dpp->samples_B [0], bptr [1]);
+ bptr [1] = sam_B;
+ dpp->samples_B [0] = bptr [0] + apply_weight (weight_A, sam_B);
+ update_weight_clip (weight_A, delta, sam_B, bptr [0]);
+ bptr [0] = dpp->samples_B [0];
+ }
+
+ break;
+
+ case -3:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_A = bptr [0] + apply_weight (weight_A, dpp->samples_A [0]);
+ update_weight_clip (weight_A, delta, dpp->samples_A [0], bptr [0]);
+ sam_B = bptr [1] + apply_weight (weight_B, dpp->samples_B [0]);
+ update_weight_clip (weight_B, delta, dpp->samples_B [0], bptr [1]);
+ bptr [0] = dpp->samples_B [0] = sam_A;
+ bptr [1] = dpp->samples_A [0] = sam_B;
+ }
+
+ break;
+ }
+
+ dpp->weight_A = weight_A;
+ dpp->weight_B = weight_B;
+}
+
+#if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM)) || defined(SIMULATOR)
+
+static void decorr_stereo_pass_cont (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
+{
+ int32_t delta = dpp->delta, weight_A = dpp->weight_A, weight_B = dpp->weight_B;
+ int32_t *bptr, *tptr, *eptr = buffer + (sample_count * 2), sam_A, sam_B;
+ int k, i;
+
+ switch (dpp->term) {
+
+ case 17:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_A = 2 * bptr [-2] - bptr [-4];
+ bptr [0] = apply_weight (weight_A, sam_A) + (sam_B = bptr [0]);
+ update_weight (weight_A, delta, sam_A, sam_B);
+
+ sam_A = 2 * bptr [-1] - bptr [-3];
+ bptr [1] = apply_weight (weight_B, sam_A) + (sam_B = bptr [1]);
+ update_weight (weight_B, delta, sam_A, sam_B);
+ }
+
+ dpp->samples_B [0] = bptr [-1];
+ dpp->samples_A [0] = bptr [-2];
+ dpp->samples_B [1] = bptr [-3];
+ dpp->samples_A [1] = bptr [-4];
+ break;
+
+ case 18:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ sam_A = (3 * bptr [-2] - bptr [-4]) >> 1;
+ bptr [0] = apply_weight (weight_A, sam_A) + (sam_B = bptr [0]);
+ update_weight (weight_A, delta, sam_A, sam_B);
+
+ sam_A = (3 * bptr [-1] - bptr [-3]) >> 1;
+ bptr [1] = apply_weight (weight_B, sam_A) + (sam_B = bptr [1]);
+ update_weight (weight_B, delta, sam_A, sam_B);
+ }
+
+ dpp->samples_B [0] = bptr [-1];
+ dpp->samples_A [0] = bptr [-2];
+ dpp->samples_B [1] = bptr [-3];
+ dpp->samples_A [1] = bptr [-4];
+ break;
+
+ default:
+ for (bptr = buffer, tptr = buffer - (dpp->term * 2); bptr < eptr; bptr += 2, tptr += 2) {
+ bptr [0] = apply_weight (weight_A, tptr [0]) + (sam_A = bptr [0]);
+ update_weight (weight_A, delta, tptr [0], sam_A);
+
+ bptr [1] = apply_weight (weight_B, tptr [1]) + (sam_A = bptr [1]);
+ update_weight (weight_B, delta, tptr [1], sam_A);
+ }
+
+ for (k = dpp->term - 1, i = 8; i--; k--) {
+ dpp->samples_B [k & (MAX_TERM - 1)] = *--bptr;
+ dpp->samples_A [k & (MAX_TERM - 1)] = *--bptr;
+ }
+
+ break;
+
+ case -1:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ bptr [0] = apply_weight (weight_A, bptr [-1]) + (sam_A = bptr [0]);
+ update_weight_clip (weight_A, delta, bptr [-1], sam_A);
+ bptr [1] = apply_weight (weight_B, bptr [0]) + (sam_A = bptr [1]);
+ update_weight_clip (weight_B, delta, bptr [0], sam_A);
+ }
+
+ dpp->samples_A [0] = bptr [-1];
+ break;
+
+ case -2:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ bptr [1] = apply_weight (weight_B, bptr [-2]) + (sam_A = bptr [1]);
+ update_weight_clip (weight_B, delta, bptr [-2], sam_A);
+ bptr [0] = apply_weight (weight_A, bptr [1]) + (sam_A = bptr [0]);
+ update_weight_clip (weight_A, delta, bptr [1], sam_A);
+ }
+
+ dpp->samples_B [0] = bptr [-2];
+ break;
+
+ case -3:
+ for (bptr = buffer; bptr < eptr; bptr += 2) {
+ bptr [0] = apply_weight (weight_A, bptr [-1]) + (sam_A = bptr [0]);
+ update_weight_clip (weight_A, delta, bptr [-1], sam_A);
+ bptr [1] = apply_weight (weight_B, bptr [-2]) + (sam_A = bptr [1]);
+ update_weight_clip (weight_B, delta, bptr [-2], sam_A);
+ }
+
+ dpp->samples_A [0] = bptr [-1];
+ dpp->samples_B [0] = bptr [-2];
+ break;
+ }
+
+ dpp->weight_A = weight_A;
+ dpp->weight_B = weight_B;
+}
+
+#endif
+
+static void decorr_mono_pass (struct decorr_pass *dpp, int32_t *buffer, int32_t sample_count)
+{
+ int32_t delta = dpp->delta, weight_A = dpp->weight_A;
+ int32_t *bptr, *eptr = buffer + sample_count, sam_A;
+ int m, k;
+
+ switch (dpp->term) {
+
+ case 17:
+ for (bptr = buffer; bptr < eptr; bptr++) {
+ sam_A = 2 * dpp->samples_A [0] - dpp->samples_A [1];
+ dpp->samples_A [1] = dpp->samples_A [0];
+ dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
+ update_weight (weight_A, delta, sam_A, bptr [0]);
+ bptr [0] = dpp->samples_A [0];
+ }
+
+ break;
+
+ case 18:
+ for (bptr = buffer; bptr < eptr; bptr++) {
+ sam_A = (3 * dpp->samples_A [0] - dpp->samples_A [1]) >> 1;
+ dpp->samples_A [1] = dpp->samples_A [0];
+ dpp->samples_A [0] = apply_weight (weight_A, sam_A) + bptr [0];
+ update_weight (weight_A, delta, sam_A, bptr [0]);
+ bptr [0] = dpp->samples_A [0];
+ }
+
+ break;
+
+ default:
+ for (m = 0, k = dpp->term & (MAX_TERM - 1), bptr = buffer; bptr < eptr; bptr++) {
+ sam_A = dpp->samples_A [m];
+ dpp->samples_A [k] = apply_weight (weight_A, sam_A) + bptr [0];
+ update_weight (weight_A, delta, sam_A, bptr [0]);
+ bptr [0] = dpp->samples_A [k];
+ m = (m + 1) & (MAX_TERM - 1);
+ k = (k + 1) & (MAX_TERM - 1);
+ }
+
+ if (m) {
+ int32_t temp_samples [MAX_TERM];
+
+ memcpy (temp_samples, dpp->samples_A, sizeof (dpp->samples_A));
+
+ for (k = 0; k < MAX_TERM; k++, m++)
+ dpp->samples_A [k] = temp_samples [m & (MAX_TERM - 1)];
+ }
+
+ break;
+ }
+
+ dpp->weight_A = weight_A;
+}
+
+
+// This is a helper function for unpack_samples() that applies several final
+// operations. First, if the data is 32-bit float data, then that conversion
+// is done in the float.c module (whether lossy or lossless) and we return.
+// Otherwise, if the extended integer data applies, then that operation is
+// executed first. If the unpacked data is lossy (and not corrected) then
+// it is clipped and shifted in a single operation. Otherwise, if it's
+// lossless then the last step is to apply the final shift (if any).
+
+static void fixup_samples (WavpackStream *wps, int32_t *buffer, uint32_t sample_count)
+{
+ uint32_t flags = wps->wphdr.flags;
+ int shift = (flags & SHIFT_MASK) >> SHIFT_LSB;
+
+ if (flags & FLOAT_DATA) {
+ float_values (wps, buffer, (flags & MONO_FLAG) ? sample_count : sample_count * 2);
+ return;
+ }
+
+ if (flags & INT32_DATA) {
+ uint32_t count = (flags & MONO_FLAG) ? sample_count : sample_count * 2;
+ int sent_bits = wps->int32_sent_bits, zeros = wps->int32_zeros;
+ int ones = wps->int32_ones, dups = wps->int32_dups;
+ int32_t *dptr = buffer;
+
+ if (!(flags & HYBRID_FLAG) && !sent_bits && (zeros + ones + dups))
+ while (count--) {
+ if (zeros)
+ *dptr <<= zeros;
+ else if (ones)
+ *dptr = ((*dptr + 1) << ones) - 1;
+ else if (dups)
+ *dptr = ((*dptr + (*dptr & 1)) << dups) - (*dptr & 1);
+
+ dptr++;
+ }
+ else
+ shift += zeros + sent_bits + ones + dups;
+ }
+
+ if (flags & HYBRID_FLAG) {
+ int32_t min_value, max_value, min_shifted, max_shifted;
+
+ switch (flags & BYTES_STORED) {
+ case 0:
+ min_shifted = (min_value = -128 >> shift) << shift;
+ max_shifted = (max_value = 127 >> shift) << shift;
+ break;
+
+ case 1:
+ min_shifted = (min_value = -32768 >> shift) << shift;
+ max_shifted = (max_value = 32767 >> shift) << shift;
+ break;
+
+ case 2:
+ min_shifted = (min_value = -8388608 >> shift) << shift;
+ max_shifted = (max_value = 8388607 >> shift) << shift;
+ break;
+
+ case 3:
+ default:
+ min_shifted = (min_value = (int32_t) 0x80000000 >> shift) << shift;
+ max_shifted = (max_value = (int32_t) 0x7FFFFFFF >> shift) << shift;
+ break;
+ }
+
+ if (!(flags & MONO_FLAG))
+ sample_count *= 2;
+
+ while (sample_count--) {
+ if (*buffer < min_value)
+ *buffer++ = min_shifted;
+ else if (*buffer > max_value)
+ *buffer++ = max_shifted;
+ else
+ *buffer++ <<= shift;
+ }
+ }
+ else if (shift) {
+ if (!(flags & MONO_FLAG))
+ sample_count *= 2;
+
+ while (sample_count--)
+ *buffer++ <<= shift;
+ }
+}
+
+// This function checks the crc value(s) for an unpacked block, returning the
+// number of actual crc errors detected for the block. The block must be
+// completely unpacked before this test is valid. For losslessly unpacked
+// blocks of float or extended integer data the extended crc is also checked.
+// Note that WavPack's crc is not a CCITT approved polynomial algorithm, but
+// is a much simpler method that is virtually as robust for real world data.
+
+int check_crc_error (WavpackContext *wpc)
+{
+ WavpackStream *wps = &wpc->stream;
+ int result = 0;
+
+ if (wps->crc != wps->wphdr.crc)
+ ++result;
+
+ return result;
+}
diff --git a/lib/wavpack/wavpack.h b/lib/wavpack/wavpack.h new file mode 100644 index 00000000..d6c35131 --- /dev/null +++ b/lib/wavpack/wavpack.h @@ -0,0 +1,394 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2004 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// wavpack.h
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include <sys/types.h>
+
+// This header file contains all the definitions required by WavPack.
+
+#ifdef __BORLANDC__
+typedef unsigned long uint32_t;
+typedef long int32_t;
+#elif defined(_WIN32) && !defined(__MINGW32__)
+#include <stdlib.h>
+typedef unsigned __int64 uint64_t;
+typedef unsigned __int32 uint32_t;
+typedef __int64 int64_t;
+typedef __int32 int32_t;
+#else
+#include <inttypes.h>
+#endif
+
+typedef unsigned char uchar;
+
+#if !defined(__GNUC__) || defined(WIN32)
+typedef unsigned short ushort;
+typedef unsigned int uint;
+#endif
+
+#include <stdio.h>
+
+#define FALSE 0
+#define TRUE 1
+
+////////////////////////////// WavPack Header /////////////////////////////////
+
+// Note that this is the ONLY structure that is written to (or read from)
+// WavPack 4.0 files, and is the preamble to every block in both the .wv
+// and .wvc files.
+
+typedef struct {
+ char ckID [4];
+ uint32_t ckSize;
+ short version;
+ uchar track_no, index_no;
+ uint32_t total_samples, block_index, block_samples, flags, crc;
+} WavpackHeader;
+
+#define WavpackHeaderFormat "4LS2LLLLL"
+
+// or-values for "flags"
+
+#define BYTES_STORED 3 // 1-4 bytes/sample
+#define MONO_FLAG 4 // not stereo
+#define HYBRID_FLAG 8 // hybrid mode
+#define JOINT_STEREO 0x10 // joint stereo
+#define CROSS_DECORR 0x20 // no-delay cross decorrelation
+#define HYBRID_SHAPE 0x40 // noise shape (hybrid mode only)
+#define FLOAT_DATA 0x80 // ieee 32-bit floating point data
+
+#define INT32_DATA 0x100 // special extended int handling
+#define HYBRID_BITRATE 0x200 // bitrate noise (hybrid mode only)
+#define HYBRID_BALANCE 0x400 // balance noise (hybrid stereo mode only)
+
+#define INITIAL_BLOCK 0x800 // initial block of multichannel segment
+#define FINAL_BLOCK 0x1000 // final block of multichannel segment
+
+#define SHIFT_LSB 13
+#define SHIFT_MASK (0x1fL << SHIFT_LSB)
+
+#define MAG_LSB 18
+#define MAG_MASK (0x1fL << MAG_LSB)
+
+#define SRATE_LSB 23
+#define SRATE_MASK (0xfL << SRATE_LSB)
+
+#define FALSE_STEREO 0x40000000 // block is stereo, but data is mono
+
+#define IGNORED_FLAGS 0x18000000 // reserved, but ignore if encountered
+#define NEW_SHAPING 0x20000000 // use IIR filter for negative shaping
+#define UNKNOWN_FLAGS 0x80000000 // also reserved, but refuse decode if
+ // encountered
+
+#define MONO_DATA (MONO_FLAG | FALSE_STEREO)
+
+#define MIN_STREAM_VERS 0x402 // lowest stream version we'll decode
+#define MAX_STREAM_VERS 0x410 // highest stream version we'll decode
+
+//////////////////////////// WavPack Metadata /////////////////////////////////
+
+// This is an internal representation of metadata.
+
+typedef struct {
+ int32_t byte_length;
+ void *data;
+ uchar id;
+} WavpackMetadata;
+
+#define ID_OPTIONAL_DATA 0x20
+#define ID_ODD_SIZE 0x40
+#define ID_LARGE 0x80
+
+#define ID_DUMMY 0x0
+#define ID_ENCODER_INFO 0x1
+#define ID_DECORR_TERMS 0x2
+#define ID_DECORR_WEIGHTS 0x3
+#define ID_DECORR_SAMPLES 0x4
+#define ID_ENTROPY_VARS 0x5
+#define ID_HYBRID_PROFILE 0x6
+#define ID_SHAPING_WEIGHTS 0x7
+#define ID_FLOAT_INFO 0x8
+#define ID_INT32_INFO 0x9
+#define ID_WV_BITSTREAM 0xa
+#define ID_WVC_BITSTREAM 0xb
+#define ID_WVX_BITSTREAM 0xc
+#define ID_CHANNEL_INFO 0xd
+
+#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1)
+#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2)
+#define ID_REPLAY_GAIN (ID_OPTIONAL_DATA | 0x3)
+#define ID_CUESHEET (ID_OPTIONAL_DATA | 0x4)
+#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5)
+#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6)
+
+///////////////////////// WavPack Configuration ///////////////////////////////
+
+// This internal structure is used during encode to provide configuration to
+// the encoding engine and during decoding to provide fle information back to
+// the higher level functions. Not all fields are used in both modes.
+
+typedef struct {
+ int bits_per_sample, bytes_per_sample;
+ int num_channels, float_norm_exp;
+ uint32_t flags, sample_rate, channel_mask;
+} WavpackConfig;
+
+#define CONFIG_BYTES_STORED 3 // 1-4 bytes/sample
+#define CONFIG_MONO_FLAG 4 // not stereo
+#define CONFIG_HYBRID_FLAG 8 // hybrid mode
+#define CONFIG_JOINT_STEREO 0x10 // joint stereo
+#define CONFIG_CROSS_DECORR 0x20 // no-delay cross decorrelation
+#define CONFIG_HYBRID_SHAPE 0x40 // noise shape (hybrid mode only)
+#define CONFIG_FLOAT_DATA 0x80 // ieee 32-bit floating point data
+
+#define CONFIG_FAST_FLAG 0x200 // fast mode
+#define CONFIG_HIGH_FLAG 0x800 // high quality mode
+#define CONFIG_VERY_HIGH_FLAG 0x1000 // very high
+#define CONFIG_BITRATE_KBPS 0x2000 // bitrate is kbps, not bits / sample
+#define CONFIG_AUTO_SHAPING 0x4000 // automatic noise shaping
+#define CONFIG_SHAPE_OVERRIDE 0x8000 // shaping mode specified
+#define CONFIG_JOINT_OVERRIDE 0x10000 // joint-stereo mode specified
+#define CONFIG_CREATE_EXE 0x40000 // create executable
+#define CONFIG_CREATE_WVC 0x80000 // create correction file
+#define CONFIG_OPTIMIZE_WVC 0x100000 // maximize bybrid compression
+#define CONFIG_CALC_NOISE 0x800000 // calc noise in hybrid mode
+#define CONFIG_LOSSY_MODE 0x1000000 // obsolete (for information)
+#define CONFIG_EXTRA_MODE 0x2000000 // extra processing mode
+#define CONFIG_SKIP_WVX 0x4000000 // no wvx stream w/ floats & big ints
+#define CONFIG_MD5_CHECKSUM 0x8000000 // compute & store MD5 signature
+#define CONFIG_OPTIMIZE_MONO 0x80000000 // optimize for mono streams posing as stereo
+
+//////////////////////////////// WavPack Stream ///////////////////////////////
+
+// This internal structure contains everything required to handle a WavPack
+// "stream", which is defined as a stereo or mono stream of audio samples. For
+// multichannel audio several of these would be required. Each stream contains
+// pointers to hold a complete allocated block of WavPack data, although it's
+// possible to decode WavPack blocks without buffering an entire block.
+
+typedef int32_t (*read_stream)(void *, void *, int32_t);
+
+typedef struct bs {
+ uchar *buf, *end, *ptr;
+ void (*wrap)(struct bs *bs);
+ uint32_t file_bytes, sr;
+ int error, bc;
+ read_stream file;
+ void *user_data;
+} Bitstream;
+
+#define MAX_NTERMS 16
+#define MAX_TERM 8
+
+struct decorr_pass {
+ short term, delta, weight_A, weight_B;
+ int32_t samples_A [MAX_TERM], samples_B [MAX_TERM];
+};
+
+struct entropy_data {
+ uint32_t median [3], slow_level, error_limit;
+};
+
+struct words_data {
+ uint32_t bitrate_delta [2], bitrate_acc [2];
+ uint32_t pend_data, holding_one, zeros_acc;
+ int holding_zero, pend_count;
+ struct entropy_data c [2];
+};
+
+typedef struct {
+ WavpackHeader wphdr;
+ Bitstream wvbits;
+
+ struct words_data w;
+
+ int num_terms, mute_error;
+ uint32_t sample_index, crc;
+
+ uchar int32_sent_bits, int32_zeros, int32_ones, int32_dups;
+ uchar float_flags, float_shift, float_max_exp, float_norm_exp;
+
+ struct decorr_pass decorr_passes [MAX_NTERMS];
+
+} WavpackStream;
+
+// flags for float_flags:
+
+#define FLOAT_SHIFT_ONES 1 // bits left-shifted into float = '1'
+#define FLOAT_SHIFT_SAME 2 // bits left-shifted into float are the same
+#define FLOAT_SHIFT_SENT 4 // bits shifted into float are sent literally
+#define FLOAT_ZEROS_SENT 8 // "zeros" are not all real zeros
+#define FLOAT_NEG_ZEROS 0x10 // contains negative zeros
+#define FLOAT_EXCEPTIONS 0x20 // contains exceptions (inf, nan, etc.)
+
+/////////////////////////////// WavPack Context ///////////////////////////////
+
+// This internal structure holds everything required to encode or decode WavPack
+// files. It is recommended that direct access to this structure be minimized
+// and the provided utilities used instead.
+
+typedef struct {
+ WavpackConfig config;
+ WavpackStream stream;
+
+ uchar read_buffer [1024];
+ char error_message [80];
+
+ read_stream infile;
+ void *user_data;
+ uint32_t total_samples, crc_errors, first_flags;
+ int open_flags, norm_offset, reduced_channels, lossy_blocks;
+
+} WavpackContext;
+
+//////////////////////// function prototypes and macros //////////////////////
+
+#define CLEAR(destin) memset (&destin, 0, sizeof (destin));
+
+// bits.c
+
+void bs_open_read (Bitstream *bs, uchar *buffer_start, uchar *buffer_end, read_stream file, void *user_data, uint32_t file_bytes);
+
+#define bs_is_open(bs) ((bs)->ptr != NULL)
+
+#define getbit(bs) ( \
+ (((bs)->bc) ? \
+ ((bs)->bc--, (bs)->sr & 1) : \
+ (((++((bs)->ptr) != (bs)->end) ? (void) 0 : (bs)->wrap (bs)), (bs)->bc = 7, ((bs)->sr = *((bs)->ptr)) & 1) \
+ ) ? \
+ ((bs)->sr >>= 1, 1) : \
+ ((bs)->sr >>= 1, 0) \
+)
+
+#define getbits(value, nbits, bs) { \
+ while ((nbits) > (bs)->bc) { \
+ if (++((bs)->ptr) == (bs)->end) (bs)->wrap (bs); \
+ (bs)->sr |= (int32_t)*((bs)->ptr) << (bs)->bc; \
+ (bs)->bc += 8; \
+ } \
+ *(value) = (bs)->sr; \
+ if ((bs)->bc > 32) { \
+ (bs)->bc -= (nbits); \
+ (bs)->sr = *((bs)->ptr) >> (8 - (bs)->bc); \
+ } \
+ else { \
+ (bs)->bc -= (nbits); \
+ (bs)->sr >>= (nbits); \
+ } \
+}
+
+void little_endian_to_native (void *data, char *format);
+void native_to_little_endian (void *data, char *format);
+
+// These macros implement the weight application and update operations
+// that are at the heart of the decorrelation loops. Note that when there
+// are several alternative versions of the same macro (marked with PERFCOND)
+// then the versions are functionally equivalent with respect to WavPack
+// decoding and the user should choose the one that provides the best
+// performance. This may be easier to check when NOT using the assembly
+// language optimizations.
+
+#if 1 // PERFCOND
+#define apply_weight_i(weight, sample) ((weight * sample + 512) >> 10)
+#else
+#define apply_weight_i(weight, sample) ((((weight * sample) >> 8) + 2) >> 2)
+#endif
+
+#define apply_weight_f(weight, sample) (((((sample & 0xffffL) * weight) >> 9) + \
+ (((sample & ~0xffffL) >> 9) * weight) + 1) >> 1)
+
+#if 0 // PERFCOND
+#define apply_weight(weight, sample) (sample != (short) sample ? \
+ apply_weight_f (weight, sample) : apply_weight_i (weight, sample))
+#else
+#define apply_weight(weight, sample) ((int32_t)((weight * (int64_t) sample + 512) >> 10))
+#endif
+
+#if 0 // PERFCOND
+#define update_weight(weight, delta, source, result) \
+ if (source && result) { int32_t s = (int32_t) (source ^ result) >> 31; weight = (delta ^ s) + (weight - s); }
+#elif 0
+#define update_weight(weight, delta, source, result) \
+ if (source && result) weight += (((source ^ result) >> 30) | 1) * delta
+#else
+#define update_weight(weight, delta, source, result) \
+ if (source && result) (source ^ result) < 0 ? (weight -= delta) : (weight += delta)
+#endif
+
+#define update_weight_clip(weight, delta, source, result) \
+ if (source && result && ((source ^ result) < 0 ? (weight -= delta) < -1024 : (weight += delta) > 1024)) \
+ weight = weight < 0 ? -1024 : 1024
+
+// unpack.c
+
+int unpack_init (WavpackContext *wpc);
+int init_wv_bitstream (WavpackContext *wpc, WavpackMetadata *wpmd);
+int read_decorr_terms (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_decorr_weights (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_decorr_samples (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_float_info (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_int32_info (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_channel_info (WavpackContext *wpc, WavpackMetadata *wpmd);
+int read_config_info (WavpackContext *wpc, WavpackMetadata *wpmd);
+int32_t unpack_samples (WavpackContext *wpc, int32_t *buffer, uint32_t sample_count);
+int check_crc_error (WavpackContext *wpc);
+
+// metadata.c stuff
+
+int read_metadata_buff (WavpackContext *wpc, WavpackMetadata *wpmd);
+int process_metadata (WavpackContext *wpc, WavpackMetadata *wpmd);
+
+// words.c stuff
+
+int read_entropy_vars (WavpackStream *wps, WavpackMetadata *wpmd);
+int read_hybrid_profile (WavpackStream *wps, WavpackMetadata *wpmd);
+int32_t get_words (int32_t *buffer, int nsamples, uint32_t flags,
+ struct words_data *w, Bitstream *bs);
+int32_t exp2s (int log);
+int restore_weight (signed char weight);
+
+#define WORD_EOF (1L << 31)
+
+// float.c
+
+int read_float_info (WavpackStream *wps, WavpackMetadata *wpmd);
+void float_values (WavpackStream *wps, int32_t *values, int32_t num_values);
+
+// wputils.c
+
+int WavpackOpenFileInput (WavpackContext *wpc, read_stream infile, void *user_data, char *error);
+
+int WavpackGetMode (WavpackContext *wpc);
+
+#define MODE_WVC 0x1
+#define MODE_LOSSLESS 0x2
+#define MODE_HYBRID 0x4
+#define MODE_FLOAT 0x8
+#define MODE_VALID_TAG 0x10
+#define MODE_HIGH 0x20
+#define MODE_FAST 0x40
+
+uint32_t WavpackUnpackSamples (WavpackContext *wpc, int32_t *buffer, uint32_t samples);
+uint32_t WavpackGetNumSamples (WavpackContext *wpc);
+uint32_t WavpackGetSampleIndex (WavpackContext *wpc);
+int WavpackGetNumErrors (WavpackContext *wpc);
+int WavpackLossyBlocks (WavpackContext *wpc);
+uint32_t WavpackGetSampleRate (WavpackContext *wpc);
+int WavpackGetBitsPerSample (WavpackContext *wpc);
+int WavpackGetBytesPerSample (WavpackContext *wpc);
+int WavpackGetNumChannels (WavpackContext *wpc);
+int WavpackGetReducedChannels (WavpackContext *wpc);
+
+#ifdef __cplusplus
+}
+#endif
diff --git a/lib/wavpack/words.c b/lib/wavpack/words.c new file mode 100644 index 00000000..0e5a3db7 --- /dev/null +++ b/lib/wavpack/words.c @@ -0,0 +1,560 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// words.c
+
+// This module provides entropy word encoding and decoding functions using
+// a variation on the Rice method. This was introduced in version 3.93
+// because it allows splitting the data into a "lossy" stream and a
+// "correction" stream in a very efficient manner and is therefore ideal
+// for the "hybrid" mode. For 4.0, the efficiency of this method was
+// significantly improved by moving away from the normal Rice restriction of
+// using powers of two for the modulus divisions and now the method can be
+// used for both hybrid and pure lossless encoding.
+
+// Samples are divided by median probabilities at 5/7 (71.43%), 10/49 (20.41%),
+// and 20/343 (5.83%). Each zone has 3.5 times fewer samples than the
+// previous. Using standard Rice coding on this data would result in 1.4
+// bits per sample average (not counting sign bit). However, there is a
+// very simple encoding that is over 99% efficient with this data and
+// results in about 1.22 bits per sample.
+
+#include "wavpack.h"
+
+#include <string.h>
+
+//////////////////////////////// local macros /////////////////////////////////
+
+#define LIMIT_ONES 16 // maximum consecutive 1s sent for "div" data
+
+// these control the time constant "slow_level" which is used for hybrid mode
+// that controls bitrate as a function of residual level (HYBRID_BITRATE).
+#define SLS 8
+#define SLO ((1 << (SLS - 1)))
+
+// these control the time constant of the 3 median level breakpoints
+#define DIV0 128 // 5/7 of samples
+#define DIV1 64 // 10/49 of samples
+#define DIV2 32 // 20/343 of samples
+
+// this macro retrieves the specified median breakpoint (without frac; min = 1)
+#define GET_MED(med) (((c->median [med]) >> 4) + 1)
+
+// These macros update the specified median breakpoints. Note that the median
+// is incremented when the sample is higher than the median, else decremented.
+// They are designed so that the median will never drop below 1 and the value
+// is essentially stationary if there are 2 increments for every 5 decrements.
+
+#define INC_MED0() (c->median [0] += ((c->median [0] + DIV0) / DIV0) * 5)
+#define DEC_MED0() (c->median [0] -= ((c->median [0] + (DIV0-2)) / DIV0) * 2)
+#define INC_MED1() (c->median [1] += ((c->median [1] + DIV1) / DIV1) * 5)
+#define DEC_MED1() (c->median [1] -= ((c->median [1] + (DIV1-2)) / DIV1) * 2)
+#define INC_MED2() (c->median [2] += ((c->median [2] + DIV2) / DIV2) * 5)
+#define DEC_MED2() (c->median [2] -= ((c->median [2] + (DIV2-2)) / DIV2) * 2)
+
+#define count_bits(av) ( \
+ (av) < (1 << 8) ? nbits_table [av] : \
+ ( \
+ (av) < (1L << 16) ? nbits_table [(av) >> 8] + 8 : \
+ ((av) < (1L << 24) ? nbits_table [(av) >> 16] + 16 : nbits_table [(av) >> 24] + 24) \
+ ) \
+)
+
+///////////////////////////// local table storage ////////////////////////////
+
+const char nbits_table [] = {
+ 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, // 0 - 15
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, // 16 - 31
+ 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, // 32 - 47
+ 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, // 48 - 63
+ 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, // 64 - 79
+ 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, // 80 - 95
+ 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, // 96 - 111
+ 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, // 112 - 127
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, // 128 - 143
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, // 144 - 159
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, // 160 - 175
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, // 176 - 191
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, // 192 - 207
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, // 208 - 223
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, // 224 - 239
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8 // 240 - 255
+};
+
+static const uchar log2_table [] = {
+ 0x00, 0x01, 0x03, 0x04, 0x06, 0x07, 0x09, 0x0a, 0x0b, 0x0d, 0x0e, 0x10, 0x11, 0x12, 0x14, 0x15,
+ 0x16, 0x18, 0x19, 0x1a, 0x1c, 0x1d, 0x1e, 0x20, 0x21, 0x22, 0x24, 0x25, 0x26, 0x28, 0x29, 0x2a,
+ 0x2c, 0x2d, 0x2e, 0x2f, 0x31, 0x32, 0x33, 0x34, 0x36, 0x37, 0x38, 0x39, 0x3b, 0x3c, 0x3d, 0x3e,
+ 0x3f, 0x41, 0x42, 0x43, 0x44, 0x45, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4d, 0x4e, 0x4f, 0x50, 0x51,
+ 0x52, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5a, 0x5c, 0x5d, 0x5e, 0x5f, 0x60, 0x61, 0x62, 0x63,
+ 0x64, 0x66, 0x67, 0x68, 0x69, 0x6a, 0x6b, 0x6c, 0x6d, 0x6e, 0x6f, 0x70, 0x71, 0x72, 0x74, 0x75,
+ 0x76, 0x77, 0x78, 0x79, 0x7a, 0x7b, 0x7c, 0x7d, 0x7e, 0x7f, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85,
+ 0x86, 0x87, 0x88, 0x89, 0x8a, 0x8b, 0x8c, 0x8d, 0x8e, 0x8f, 0x90, 0x91, 0x92, 0x93, 0x94, 0x95,
+ 0x96, 0x97, 0x98, 0x99, 0x9a, 0x9b, 0x9b, 0x9c, 0x9d, 0x9e, 0x9f, 0xa0, 0xa1, 0xa2, 0xa3, 0xa4,
+ 0xa5, 0xa6, 0xa7, 0xa8, 0xa9, 0xa9, 0xaa, 0xab, 0xac, 0xad, 0xae, 0xaf, 0xb0, 0xb1, 0xb2, 0xb2,
+ 0xb3, 0xb4, 0xb5, 0xb6, 0xb7, 0xb8, 0xb9, 0xb9, 0xba, 0xbb, 0xbc, 0xbd, 0xbe, 0xbf, 0xc0, 0xc0,
+ 0xc1, 0xc2, 0xc3, 0xc4, 0xc5, 0xc6, 0xc6, 0xc7, 0xc8, 0xc9, 0xca, 0xcb, 0xcb, 0xcc, 0xcd, 0xce,
+ 0xcf, 0xd0, 0xd0, 0xd1, 0xd2, 0xd3, 0xd4, 0xd4, 0xd5, 0xd6, 0xd7, 0xd8, 0xd8, 0xd9, 0xda, 0xdb,
+ 0xdc, 0xdc, 0xdd, 0xde, 0xdf, 0xe0, 0xe0, 0xe1, 0xe2, 0xe3, 0xe4, 0xe4, 0xe5, 0xe6, 0xe7, 0xe7,
+ 0xe8, 0xe9, 0xea, 0xea, 0xeb, 0xec, 0xed, 0xee, 0xee, 0xef, 0xf0, 0xf1, 0xf1, 0xf2, 0xf3, 0xf4,
+ 0xf4, 0xf5, 0xf6, 0xf7, 0xf7, 0xf8, 0xf9, 0xf9, 0xfa, 0xfb, 0xfc, 0xfc, 0xfd, 0xfe, 0xff, 0xff
+};
+
+static const uchar exp2_table [] = {
+ 0x00, 0x01, 0x01, 0x02, 0x03, 0x03, 0x04, 0x05, 0x06, 0x06, 0x07, 0x08, 0x08, 0x09, 0x0a, 0x0b,
+ 0x0b, 0x0c, 0x0d, 0x0e, 0x0e, 0x0f, 0x10, 0x10, 0x11, 0x12, 0x13, 0x13, 0x14, 0x15, 0x16, 0x16,
+ 0x17, 0x18, 0x19, 0x19, 0x1a, 0x1b, 0x1c, 0x1d, 0x1d, 0x1e, 0x1f, 0x20, 0x20, 0x21, 0x22, 0x23,
+ 0x24, 0x24, 0x25, 0x26, 0x27, 0x28, 0x28, 0x29, 0x2a, 0x2b, 0x2c, 0x2c, 0x2d, 0x2e, 0x2f, 0x30,
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3a, 0x3a, 0x3b, 0x3c, 0x3d,
+ 0x3e, 0x3f, 0x40, 0x41, 0x41, 0x42, 0x43, 0x44, 0x45, 0x46, 0x47, 0x48, 0x48, 0x49, 0x4a, 0x4b,
+ 0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5a,
+ 0x5b, 0x5c, 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66, 0x67, 0x68, 0x69,
+ 0x6a, 0x6b, 0x6c, 0x6d, 0x6e, 0x6f, 0x70, 0x71, 0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79,
+ 0x7a, 0x7b, 0x7c, 0x7d, 0x7e, 0x7f, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x87, 0x88, 0x89, 0x8a,
+ 0x8b, 0x8c, 0x8d, 0x8e, 0x8f, 0x90, 0x91, 0x92, 0x93, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9a, 0x9b,
+ 0x9c, 0x9d, 0x9f, 0xa0, 0xa1, 0xa2, 0xa3, 0xa4, 0xa5, 0xa6, 0xa8, 0xa9, 0xaa, 0xab, 0xac, 0xad,
+ 0xaf, 0xb0, 0xb1, 0xb2, 0xb3, 0xb4, 0xb6, 0xb7, 0xb8, 0xb9, 0xba, 0xbc, 0xbd, 0xbe, 0xbf, 0xc0,
+ 0xc2, 0xc3, 0xc4, 0xc5, 0xc6, 0xc8, 0xc9, 0xca, 0xcb, 0xcd, 0xce, 0xcf, 0xd0, 0xd2, 0xd3, 0xd4,
+ 0xd6, 0xd7, 0xd8, 0xd9, 0xdb, 0xdc, 0xdd, 0xde, 0xe0, 0xe1, 0xe2, 0xe4, 0xe5, 0xe6, 0xe8, 0xe9,
+ 0xea, 0xec, 0xed, 0xee, 0xf0, 0xf1, 0xf2, 0xf4, 0xf5, 0xf6, 0xf8, 0xf9, 0xfa, 0xfc, 0xfd, 0xff
+};
+
+static const char ones_count_table [] = {
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,6,
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,7,
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,6,
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,5,
+ 0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,4,0,1,0,2,0,1,0,3,0,1,0,2,0,1,0,8
+};
+
+///////////////////////////// executable code ////////////////////////////////
+
+void init_words (WavpackStream *wps)
+{
+ CLEAR (wps->w);
+}
+
+static int mylog2 (uint32_t avalue);
+
+// Read the median log2 values from the specifed metadata structure, convert
+// them back to 32-bit unsigned values and store them. If length is not
+// exactly correct then we flag and return an error.
+
+int read_entropy_vars (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+ uchar *byteptr = wpmd->data;
+
+ if (wpmd->byte_length != ((wps->wphdr.flags & MONO_DATA) ? 6 : 12))
+ return FALSE;
+
+ wps->w.c [0].median [0] = exp2s (byteptr [0] + (byteptr [1] << 8));
+ wps->w.c [0].median [1] = exp2s (byteptr [2] + (byteptr [3] << 8));
+ wps->w.c [0].median [2] = exp2s (byteptr [4] + (byteptr [5] << 8));
+
+ if (!(wps->wphdr.flags & MONO_DATA)) {
+ wps->w.c [1].median [0] = exp2s (byteptr [6] + (byteptr [7] << 8));
+ wps->w.c [1].median [1] = exp2s (byteptr [8] + (byteptr [9] << 8));
+ wps->w.c [1].median [2] = exp2s (byteptr [10] + (byteptr [11] << 8));
+ }
+
+ return TRUE;
+}
+
+// Read the hybrid related values from the specifed metadata structure, convert
+// them back to their internal formats and store them. The extended profile
+// stuff is not implemented yet, so return an error if we get more data than
+// we know what to do with.
+
+int read_hybrid_profile (WavpackStream *wps, WavpackMetadata *wpmd)
+{
+ uchar *byteptr = wpmd->data;
+ uchar *endptr = byteptr + wpmd->byte_length;
+
+ if (wps->wphdr.flags & HYBRID_BITRATE) {
+ wps->w.c [0].slow_level = exp2s (byteptr [0] + (byteptr [1] << 8));
+ byteptr += 2;
+
+ if (!(wps->wphdr.flags & MONO_DATA)) {
+ wps->w.c [1].slow_level = exp2s (byteptr [0] + (byteptr [1] << 8));
+ byteptr += 2;
+ }
+ }
+
+ wps->w.bitrate_acc [0] = (int32_t)(byteptr [0] + (byteptr [1] << 8)) << 16;
+ byteptr += 2;
+
+ if (!(wps->wphdr.flags & MONO_DATA)) {
+ wps->w.bitrate_acc [1] = (int32_t)(byteptr [0] + (byteptr [1] << 8)) << 16;
+ byteptr += 2;
+ }
+
+ if (byteptr < endptr) {
+ wps->w.bitrate_delta [0] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+ byteptr += 2;
+
+ if (!(wps->wphdr.flags & MONO_DATA)) {
+ wps->w.bitrate_delta [1] = exp2s ((short)(byteptr [0] + (byteptr [1] << 8)));
+ byteptr += 2;
+ }
+
+ if (byteptr < endptr)
+ return FALSE;
+ }
+ else
+ wps->w.bitrate_delta [0] = wps->w.bitrate_delta [1] = 0;
+
+ return TRUE;
+}
+
+// This function is called during both encoding and decoding of hybrid data to
+// update the "error_limit" variable which determines the maximum sample error
+// allowed in the main bitstream. In the HYBRID_BITRATE mode (which is the only
+// currently implemented) this is calculated from the slow_level values and the
+// bitrate accumulators. Note that the bitrate accumulators can be changing.
+
+void update_error_limit (struct words_data *w, uint32_t flags)
+{
+ int bitrate_0 = (w->bitrate_acc [0] += w->bitrate_delta [0]) >> 16;
+
+ if (flags & MONO_DATA) {
+ if (flags & HYBRID_BITRATE) {
+ int slow_log_0 = (w->c [0].slow_level + SLO) >> SLS;
+
+ if (slow_log_0 - bitrate_0 > -0x100)
+ w->c [0].error_limit = exp2s (slow_log_0 - bitrate_0 + 0x100);
+ else
+ w->c [0].error_limit = 0;
+ }
+ else
+ w->c [0].error_limit = exp2s (bitrate_0);
+ }
+ else {
+ int bitrate_1 = (w->bitrate_acc [1] += w->bitrate_delta [1]) >> 16;
+
+ if (flags & HYBRID_BITRATE) {
+ int slow_log_0 = (w->c [0].slow_level + SLO) >> SLS;
+ int slow_log_1 = (w->c [1].slow_level + SLO) >> SLS;
+
+ if (flags & HYBRID_BALANCE) {
+ int balance = (slow_log_1 - slow_log_0 + bitrate_1 + 1) >> 1;
+
+ if (balance > bitrate_0) {
+ bitrate_1 = bitrate_0 * 2;
+ bitrate_0 = 0;
+ }
+ else if (-balance > bitrate_0) {
+ bitrate_0 = bitrate_0 * 2;
+ bitrate_1 = 0;
+ }
+ else {
+ bitrate_1 = bitrate_0 + balance;
+ bitrate_0 = bitrate_0 - balance;
+ }
+ }
+
+ if (slow_log_0 - bitrate_0 > -0x100)
+ w->c [0].error_limit = exp2s (slow_log_0 - bitrate_0 + 0x100);
+ else
+ w->c [0].error_limit = 0;
+
+ if (slow_log_1 - bitrate_1 > -0x100)
+ w->c [1].error_limit = exp2s (slow_log_1 - bitrate_1 + 0x100);
+ else
+ w->c [1].error_limit = 0;
+ }
+ else {
+ w->c [0].error_limit = exp2s (bitrate_0);
+ w->c [1].error_limit = exp2s (bitrate_1);
+ }
+ }
+}
+
+static uint32_t read_code (Bitstream *bs, uint32_t maxcode);
+
+// Read the next word from the bitstream "wvbits" and return the value. This
+// function can be used for hybrid or lossless streams, but since an
+// optimized version is available for lossless this function would normally
+// be used for hybrid only. If a hybrid lossless stream is being read then
+// the "correction" offset is written at the specified pointer. A return value
+// of WORD_EOF indicates that the end of the bitstream was reached (all 1s) or
+// some other error occurred.
+
+int32_t get_words (int32_t *buffer, int nsamples, uint32_t flags,
+ struct words_data *w, Bitstream *bs)
+{
+ register struct entropy_data *c = w->c;
+ int csamples;
+
+ if (!(flags & MONO_DATA))
+ nsamples *= 2;
+
+ for (csamples = 0; csamples < nsamples; ++csamples) {
+ uint32_t ones_count, low, mid, high;
+
+ if (!(flags & MONO_DATA))
+ c = w->c + (csamples & 1);
+
+ if (!(w->c [0].median [0] & ~1) && !w->holding_zero && !w->holding_one && !(w->c [1].median [0] & ~1)) {
+ uint32_t mask;
+ int cbits;
+
+ if (w->zeros_acc) {
+ if (--w->zeros_acc) {
+ c->slow_level -= (c->slow_level + SLO) >> SLS;
+ *buffer++ = 0;
+ continue;
+ }
+ }
+ else {
+ for (cbits = 0; cbits < 33 && getbit (bs); ++cbits);
+
+ if (cbits == 33)
+ break;
+
+ if (cbits < 2)
+ w->zeros_acc = cbits;
+ else {
+ for (mask = 1, w->zeros_acc = 0; --cbits; mask <<= 1)
+ if (getbit (bs))
+ w->zeros_acc |= mask;
+
+ w->zeros_acc |= mask;
+ }
+
+ if (w->zeros_acc) {
+ c->slow_level -= (c->slow_level + SLO) >> SLS;
+ CLEAR (w->c [0].median);
+ CLEAR (w->c [1].median);
+ *buffer++ = 0;
+ continue;
+ }
+ }
+ }
+
+ if (w->holding_zero)
+ ones_count = w->holding_zero = 0;
+ else {
+ int next8;
+
+ if (bs->bc < 8) {
+ if (++(bs->ptr) == bs->end)
+ bs->wrap (bs);
+
+ next8 = (bs->sr |= *(bs->ptr) << bs->bc) & 0xff;
+ bs->bc += 8;
+ }
+ else
+ next8 = bs->sr & 0xff;
+
+ if (next8 == 0xff) {
+ bs->bc -= 8;
+ bs->sr >>= 8;
+
+ for (ones_count = 8; ones_count < (LIMIT_ONES + 1) && getbit (bs); ++ones_count);
+
+ if (ones_count == (LIMIT_ONES + 1))
+ break;
+
+ if (ones_count == LIMIT_ONES) {
+ uint32_t mask;
+ int cbits;
+
+ for (cbits = 0; cbits < 33 && getbit (bs); ++cbits);
+
+ if (cbits == 33)
+ break;
+
+ if (cbits < 2)
+ ones_count = cbits;
+ else {
+ for (mask = 1, ones_count = 0; --cbits; mask <<= 1)
+ if (getbit (bs))
+ ones_count |= mask;
+
+ ones_count |= mask;
+ }
+
+ ones_count += LIMIT_ONES;
+ }
+ }
+ else {
+ bs->bc -= (ones_count = ones_count_table [next8]) + 1;
+ bs->sr >>= ones_count + 1;
+ }
+
+ if (w->holding_one) {
+ w->holding_one = ones_count & 1;
+ ones_count = (ones_count >> 1) + 1;
+ }
+ else {
+ w->holding_one = ones_count & 1;
+ ones_count >>= 1;
+ }
+
+ w->holding_zero = ~w->holding_one & 1;
+ }
+
+ if ((flags & HYBRID_FLAG) && ((flags & MONO_DATA) || !(csamples & 1)))
+ update_error_limit (w, flags);
+
+ if (ones_count == 0) {
+ low = 0;
+ high = GET_MED (0) - 1;
+ DEC_MED0 ();
+ }
+ else {
+ low = GET_MED (0);
+ INC_MED0 ();
+
+ if (ones_count == 1) {
+ high = low + GET_MED (1) - 1;
+ DEC_MED1 ();
+ }
+ else {
+ low += GET_MED (1);
+ INC_MED1 ();
+
+ if (ones_count == 2) {
+ high = low + GET_MED (2) - 1;
+ DEC_MED2 ();
+ }
+ else {
+ low += (ones_count - 2) * GET_MED (2);
+ high = low + GET_MED (2) - 1;
+ INC_MED2 ();
+ }
+ }
+ }
+
+ mid = (high + low + 1) >> 1;
+
+ if (!c->error_limit)
+ mid = read_code (bs, high - low) + low;
+ else while (high - low > c->error_limit) {
+ if (getbit (bs))
+ mid = (high + (low = mid) + 1) >> 1;
+ else
+ mid = ((high = mid - 1) + low + 1) >> 1;
+ }
+
+ *buffer++ = getbit (bs) ? ~mid : mid;
+
+ if (flags & HYBRID_BITRATE)
+ c->slow_level = c->slow_level - ((c->slow_level + SLO) >> SLS) + mylog2 (mid);
+ }
+
+ return (flags & MONO_DATA) ? csamples : (csamples / 2);
+}
+
+// Read a single unsigned value from the specified bitstream with a value
+// from 0 to maxcode. If there are exactly a power of two number of possible
+// codes then this will read a fixed number of bits; otherwise it reads the
+// minimum number of bits and then determines whether another bit is needed
+// to define the code.
+
+static uint32_t read_code (Bitstream *bs, uint32_t maxcode)
+{
+ int bitcount = count_bits (maxcode);
+ uint32_t extras = (1L << bitcount) - maxcode - 1, code;
+
+ if (!bitcount)
+ return 0;
+
+ getbits (&code, bitcount - 1, bs);
+ code &= (1L << (bitcount - 1)) - 1;
+
+ if (code >= extras) {
+ code = (code << 1) - extras;
+
+ if (getbit (bs))
+ ++code;
+ }
+
+ return code;
+}
+
+// The concept of a base 2 logarithm is used in many parts of WavPack. It is
+// a way of sufficiently accurately representing 32-bit signed and unsigned
+// values storing only 16 bits (actually fewer). It is also used in the hybrid
+// mode for quickly comparing the relative magnitude of large values (i.e.
+// division) and providing smooth exponentials using only addition.
+
+// These are not strict logarithms in that they become linear around zero and
+// can therefore represent both zero and negative values. They have 8 bits
+// of precision and in "roundtrip" conversions the total error never exceeds 1
+// part in 225 except for the cases of +/-115 and +/-195 (which error by 1).
+
+
+// This function returns the log2 for the specified 32-bit unsigned value.
+// The maximum value allowed is about 0xff800000 and returns 8447.
+
+static int mylog2 (uint32_t avalue)
+{
+ int dbits;
+
+ if ((avalue += avalue >> 9) < (1 << 8)) {
+ dbits = nbits_table [avalue];
+ return (dbits << 8) + log2_table [(avalue << (9 - dbits)) & 0xff];
+ }
+ else {
+ if (avalue < (1L << 16))
+ dbits = nbits_table [avalue >> 8] + 8;
+ else if (avalue < (1L << 24))
+ dbits = nbits_table [avalue >> 16] + 16;
+ else
+ dbits = nbits_table [avalue >> 24] + 24;
+
+ return (dbits << 8) + log2_table [(avalue >> (dbits - 9)) & 0xff];
+ }
+}
+
+// This function returns the log2 for the specified 32-bit signed value.
+// All input values are valid and the return values are in the range of
+// +/- 8192.
+
+int log2s (int32_t value)
+{
+ return (value < 0) ? -mylog2 (-value) : mylog2 (value);
+}
+
+// This function returns the original integer represented by the supplied
+// logarithm (at least within the provided accuracy). The log is signed,
+// but since a full 32-bit value is returned this can be used for unsigned
+// conversions as well (i.e. the input range is -8192 to +8447).
+
+int32_t exp2s (int log)
+{
+ uint32_t value;
+
+ if (log < 0)
+ return -exp2s (-log);
+
+ value = exp2_table [log & 0xff] | 0x100;
+
+ if ((log >>= 8) <= 9)
+ return value >> (9 - log);
+ else
+ return value << (log - 9);
+}
+
+// These two functions convert internal weights (which are normally +/-1024)
+// to and from an 8-bit signed character version for storage in metadata. The
+// weights are clipped here in the case that they are outside that range.
+
+int restore_weight (signed char weight)
+{
+ int result;
+
+ if ((result = (int) weight << 3) > 0)
+ result += (result + 64) >> 7;
+
+ return result;
+}
diff --git a/lib/wavpack/wputils.c b/lib/wavpack/wputils.c new file mode 100644 index 00000000..c3a23f98 --- /dev/null +++ b/lib/wavpack/wputils.c @@ -0,0 +1,350 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// wputils.c
+
+// This module provides a high-level interface for decoding WavPack 4.0 audio
+// streams and files. WavPack data is read with a stream reading callback. No
+// direct seeking is provided for, but it is possible to start decoding
+// anywhere in a WavPack stream. In this case, WavPack will be able to provide
+// the sample-accurate position when it synchs with the data and begins
+// decoding.
+
+#include "wavpack.h"
+
+#include <string.h>
+
+///////////////////////////// local table storage ////////////////////////////
+
+const uint32_t sample_rates [] = { 6000, 8000, 9600, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000 };
+
+///////////////////////////// executable code ////////////////////////////////
+
+static uint32_t read_next_header (read_stream infile, void *user_data, WavpackHeader *wphdr);
+
+// This function reads data from the specified stream in search of a valid
+// WavPack 4.0 audio block. If this fails in 1 megabyte (or an invalid or
+// unsupported WavPack block is encountered) then an appropriate message is
+// copied to "error" and NULL is returned, otherwise a pointer to a
+// WavpackContext structure is returned (which is used to call all other
+// functions in this module). This can be initiated at the beginning of a
+// WavPack file, or anywhere inside a WavPack file. To determine the exact
+// position within the file use WavpackGetSampleIndex(). For demonstration
+// purposes this uses a single static copy of the WavpackContext structure,
+// so obviously it cannot be used for more than one file at a time. Also,
+// this function will not handle "correction" files, plays only the first
+// two channels of multi-channel files, and is limited in resolution in some
+// large integer or floating point files (but always provides at least 24 bits
+// of resolution).
+
+int WavpackOpenFileInput (WavpackContext *wpc, read_stream infile, void *user_data, char *error)
+{
+ WavpackStream *wps = &wpc->stream;
+ uint32_t bcount;
+
+ //CLEAR (wpc);
+ wpc->infile = infile;
+ wpc->user_data = user_data;
+ wpc->total_samples = (uint32_t) -1;
+ wpc->norm_offset = 0;
+ wpc->open_flags = 0;
+
+ // open the source file for reading and store the size
+
+ while (!wps->wphdr.block_samples) {
+
+ bcount = read_next_header (wpc->infile, wpc->user_data, &wps->wphdr);
+
+ if (bcount == (uint32_t) -1) {
+ strcpy (error, "not compatible with this version of WavPack file!");
+ return 0;
+ }
+
+ if (wps->wphdr.block_samples && wps->wphdr.total_samples != (uint32_t) -1)
+ wpc->total_samples = wps->wphdr.total_samples;
+
+ if (!unpack_init (wpc)) {
+ strcpy (error, wpc->error_message [0] ? wpc->error_message :
+ "not compatible with this version of WavPack file!");
+
+ return 0;
+ }
+ }
+
+ wpc->config.flags &= ~0xff;
+ wpc->config.flags |= wps->wphdr.flags & 0xff;
+ wpc->config.bytes_per_sample = (wps->wphdr.flags & BYTES_STORED) + 1;
+ wpc->config.float_norm_exp = wps->float_norm_exp;
+
+ wpc->config.bits_per_sample = (wpc->config.bytes_per_sample * 8) -
+ ((wps->wphdr.flags & SHIFT_MASK) >> SHIFT_LSB);
+
+ if (wpc->config.flags & FLOAT_DATA) {
+ wpc->config.bytes_per_sample = 3;
+ wpc->config.bits_per_sample = 24;
+ }
+
+ if (!wpc->config.sample_rate) {
+ if (!wps || !wps->wphdr.block_samples || (wps->wphdr.flags & SRATE_MASK) == SRATE_MASK)
+ wpc->config.sample_rate = 44100;
+ else
+ wpc->config.sample_rate = sample_rates [(wps->wphdr.flags & SRATE_MASK) >> SRATE_LSB];
+ }
+
+ if (!wpc->config.num_channels) {
+ wpc->config.num_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
+ wpc->config.channel_mask = 0x5 - wpc->config.num_channels;
+ }
+
+ if (!(wps->wphdr.flags & FINAL_BLOCK))
+ wpc->reduced_channels = (wps->wphdr.flags & MONO_FLAG) ? 1 : 2;
+
+ return 1;
+}
+
+// This function obtains general information about an open file and returns
+// a mask with the following bit values:
+
+// MODE_LOSSLESS: file is lossless (pure lossless only)
+// MODE_HYBRID: file is hybrid mode (lossy part only)
+// MODE_FLOAT: audio data is 32-bit ieee floating point (but will provided
+// in 24-bit integers for convenience)
+// MODE_HIGH: file was created in "high" mode (information only)
+// MODE_FAST: file was created in "fast" mode (information only)
+
+int WavpackGetMode (WavpackContext *wpc)
+{
+ int mode = 0;
+
+ if (wpc) {
+ if (wpc->config.flags & CONFIG_HYBRID_FLAG)
+ mode |= MODE_HYBRID;
+ else if (!(wpc->config.flags & CONFIG_LOSSY_MODE))
+ mode |= MODE_LOSSLESS;
+
+ if (wpc->lossy_blocks)
+ mode &= ~MODE_LOSSLESS;
+
+ if (wpc->config.flags & CONFIG_FLOAT_DATA)
+ mode |= MODE_FLOAT;
+
+ if (wpc->config.flags & CONFIG_HIGH_FLAG)
+ mode |= MODE_HIGH;
+
+ if (wpc->config.flags & CONFIG_FAST_FLAG)
+ mode |= MODE_FAST;
+ }
+
+ return mode;
+}
+
+// Unpack the specified number of samples from the current file position.
+// Note that "samples" here refers to "complete" samples, which would be
+// 2 longs for stereo files. The audio data is returned right-justified in
+// 32-bit longs in the endian mode native to the executing processor. So,
+// if the original data was 16-bit, then the values returned would be
+// +/-32k. Floating point data will be returned as 24-bit integers (and may
+// also be clipped). The actual number of samples unpacked is returned,
+// which should be equal to the number requested unless the end of fle is
+// encountered or an error occurs.
+
+uint32_t WavpackUnpackSamples (WavpackContext *wpc, int32_t *buffer, uint32_t samples)
+{
+ WavpackStream *wps = &wpc->stream;
+ uint32_t bcount, samples_unpacked = 0, samples_to_unpack;
+ int num_channels = wpc->config.num_channels;
+
+ while (samples) {
+ if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
+ wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples) {
+ bcount = read_next_header (wpc->infile, wpc->user_data, &wps->wphdr);
+
+ if (bcount == (uint32_t) -1)
+ break;
+
+ if (!wps->wphdr.block_samples || wps->sample_index == wps->wphdr.block_index)
+ if (!unpack_init (wpc))
+ break;
+ }
+
+ if (!wps->wphdr.block_samples || !(wps->wphdr.flags & INITIAL_BLOCK) ||
+ wps->sample_index >= wps->wphdr.block_index + wps->wphdr.block_samples)
+ continue;
+
+ if (wps->sample_index < wps->wphdr.block_index) {
+ samples_to_unpack = wps->wphdr.block_index - wps->sample_index;
+
+ if (samples_to_unpack > samples)
+ samples_to_unpack = samples;
+
+ wps->sample_index += samples_to_unpack;
+ samples_unpacked += samples_to_unpack;
+ samples -= samples_to_unpack;
+
+ if (wpc->reduced_channels)
+ samples_to_unpack *= wpc->reduced_channels;
+ else
+ samples_to_unpack *= num_channels;
+
+ while (samples_to_unpack--)
+ *buffer++ = 0;
+
+ continue;
+ }
+
+ samples_to_unpack = wps->wphdr.block_index + wps->wphdr.block_samples - wps->sample_index;
+
+ if (samples_to_unpack > samples)
+ samples_to_unpack = samples;
+
+ unpack_samples (wpc, buffer, samples_to_unpack);
+
+ if (wpc->reduced_channels)
+ buffer += samples_to_unpack * wpc->reduced_channels;
+ else
+ buffer += samples_to_unpack * num_channels;
+
+ samples_unpacked += samples_to_unpack;
+ samples -= samples_to_unpack;
+
+ if (wps->sample_index == wps->wphdr.block_index + wps->wphdr.block_samples) {
+ if (check_crc_error (wpc))
+ wpc->crc_errors++;
+ }
+
+ if (wps->sample_index == wpc->total_samples)
+ break;
+ }
+
+ return samples_unpacked;
+}
+
+// Get total number of samples contained in the WavPack file, or -1 if unknown
+
+uint32_t WavpackGetNumSamples (WavpackContext *wpc)
+{
+ return wpc ? wpc->total_samples : (uint32_t) -1;
+}
+
+// Get the current sample index position, or -1 if unknown
+
+uint32_t WavpackGetSampleIndex (WavpackContext *wpc)
+{
+ if (wpc)
+ return wpc->stream.sample_index;
+
+ return (uint32_t) -1;
+}
+
+// Get the number of errors encountered so far
+
+int WavpackGetNumErrors (WavpackContext *wpc)
+{
+ return wpc ? wpc->crc_errors : 0;
+}
+
+// return TRUE if any uncorrected lossy blocks were actually written or read
+
+int WavpackLossyBlocks (WavpackContext *wpc)
+{
+ return wpc ? wpc->lossy_blocks : 0;
+}
+
+// Returns the sample rate of the specified WavPack file
+
+uint32_t WavpackGetSampleRate (WavpackContext *wpc)
+{
+ return wpc ? wpc->config.sample_rate : 44100;
+}
+
+// Returns the number of channels of the specified WavPack file. Note that
+// this is the actual number of channels contained in the file, but this
+// version can only decode the first two.
+
+int WavpackGetNumChannels (WavpackContext *wpc)
+{
+ return wpc ? wpc->config.num_channels : 2;
+}
+
+// Returns the actual number of valid bits per sample contained in the
+// original file, which may or may not be a multiple of 8. Floating data
+// always has 32 bits, integers may be from 1 to 32 bits each. When this
+// value is not a multiple of 8, then the "extra" bits are located in the
+// LSBs of the results. That is, values are right justified when unpacked
+// into longs, but are left justified in the number of bytes used by the
+// original data.
+
+int WavpackGetBitsPerSample (WavpackContext *wpc)
+{
+ return wpc ? wpc->config.bits_per_sample : 16;
+}
+
+// Returns the number of bytes used for each sample (1 to 4) in the original
+// file. This is required information for the user of this module because the
+// audio data is returned in the LOWER bytes of the long buffer and must be
+// left-shifted 8, 16, or 24 bits if normalized longs are required.
+
+int WavpackGetBytesPerSample (WavpackContext *wpc)
+{
+ return wpc ? wpc->config.bytes_per_sample : 2;
+}
+
+// This function will return the actual number of channels decoded from the
+// file (which may or may not be less than the actual number of channels, but
+// will always be 1 or 2). Normally, this will be the front left and right
+// channels of a multi-channel file.
+
+int WavpackGetReducedChannels (WavpackContext *wpc)
+{
+ if (wpc)
+ return wpc->reduced_channels ? wpc->reduced_channels : wpc->config.num_channels;
+ else
+ return 2;
+}
+
+// Read from current file position until a valid 32-byte WavPack 4.0 header is
+// found and read into the specified pointer. The number of bytes skipped is
+// returned. If no WavPack header is found within 1 meg, then a -1 is returned
+// to indicate the error. No additional bytes are read past the header and it
+// is returned in the processor's native endian mode. Seeking is not required.
+
+static uint32_t read_next_header (read_stream infile, void *user_data, WavpackHeader *wphdr)
+{
+ char buffer [sizeof (*wphdr)], *sp = buffer + sizeof (*wphdr), *ep = sp;
+ uint32_t bytes_skipped = 0;
+ int bleft;
+
+ while (1) {
+ if (sp < ep) {
+ bleft = ep - sp;
+ memcpy (buffer, sp, bleft);
+ }
+ else
+ bleft = 0;
+
+ if (infile (user_data, buffer + bleft, sizeof (*wphdr) - bleft) != (int32_t) sizeof (*wphdr) - bleft)
+ return -1;
+
+ sp = buffer;
+
+ if (*sp++ == 'w' && *sp == 'v' && *++sp == 'p' && *++sp == 'k' &&
+ !(*++sp & 1) && sp [2] < 16 && !sp [3] && sp [5] == 4 &&
+ sp [4] >= (MIN_STREAM_VERS & 0xff) && sp [4] <= (MAX_STREAM_VERS & 0xff)) {
+ memcpy (wphdr, buffer, sizeof (*wphdr));
+ little_endian_to_native (wphdr, WavpackHeaderFormat);
+ return bytes_skipped;
+ }
+
+ while (sp < ep && *sp != 'w')
+ sp++;
+
+ if ((bytes_skipped += sp - buffer) > 1048576L)
+ return -1;
+ }
+}
diff --git a/lib/wavpack/wvfilter.c b/lib/wavpack/wvfilter.c new file mode 100644 index 00000000..f80d73dd --- /dev/null +++ b/lib/wavpack/wvfilter.c @@ -0,0 +1,200 @@ +////////////////////////////////////////////////////////////////////////////
+// **** WAVPACK **** //
+// Hybrid Lossless Wavefile Compressor //
+// Copyright (c) 1998 - 2006 Conifer Software. //
+// All Rights Reserved. //
+// Distributed under the BSD Software License (see license.txt) //
+////////////////////////////////////////////////////////////////////////////
+
+// wv_filter.c
+
+// This is the main module for the demonstration WavPack command-line
+// decoder filter. It uses the tiny "hardware" version of the decoder and
+// accepts WavPack files on stdin and outputs a standard MS wav file to
+// stdout. Note that this involves converting the data to little-endian
+// (if the executing processor is not), possibly packing the data into
+// fewer bytes per sample, and generating an appropriate riff wav header.
+// Note that this is NOT the copy of the RIFF header that might be stored
+// in the file, and any additional RIFF information and tags are lost.
+// See wputils.c for further limitations.
+
+#include "wavpack.h"
+
+#if defined(WIN32)
+#include <io.h>
+#include <fcntl.h>
+#endif
+
+#include <string.h>
+
+// These structures are used to place a wav riff header at the beginning of
+// the output.
+
+typedef struct {
+ char ckID [4];
+ uint32_t ckSize;
+ char formType [4];
+} RiffChunkHeader;
+
+typedef struct {
+ char ckID [4];
+ uint32_t ckSize;
+} ChunkHeader;
+
+#define ChunkHeaderFormat "4L"
+
+typedef struct {
+ ushort FormatTag, NumChannels;
+ uint32_t SampleRate, BytesPerSecond;
+ ushort BlockAlign, BitsPerSample;
+} WaveHeader;
+
+#define WaveHeaderFormat "SSLLSS"
+
+static uchar *format_samples (int bps, uchar *dst, int32_t *src, uint32_t samcnt);
+static int32_t read_bytes (void *buff, int32_t bcount);
+static int32_t temp_buffer [256];
+
+int main ()
+{
+ ChunkHeader FormatChunkHeader, DataChunkHeader;
+ RiffChunkHeader RiffChunkHeader;
+ WaveHeader WaveHeader;
+
+ uint32_t total_unpacked_samples = 0, total_samples;
+ int num_channels, bps;
+ WavpackContext *wpc;
+ char error [80];
+
+#if defined(WIN32)
+ setmode (fileno (stdin), O_BINARY);
+ setmode (fileno (stdout), O_BINARY);
+#endif
+
+ wpc = WavpackOpenFileInput (read_bytes, error);
+
+ if (!wpc) {
+ fputs (error, stderr);
+ fputs ("\n", stderr);
+ return 1;
+ }
+
+ num_channels = WavpackGetReducedChannels (wpc);
+ total_samples = WavpackGetNumSamples (wpc);
+ bps = WavpackGetBytesPerSample (wpc);
+
+ strncpy (RiffChunkHeader.ckID, "RIFF", sizeof (RiffChunkHeader.ckID));
+ RiffChunkHeader.ckSize = total_samples * num_channels * bps + sizeof (ChunkHeader) * 2 + sizeof (WaveHeader) + 4;
+ strncpy (RiffChunkHeader.formType, "WAVE", sizeof (RiffChunkHeader.formType));
+
+ strncpy (FormatChunkHeader.ckID, "fmt ", sizeof (FormatChunkHeader.ckID));
+ FormatChunkHeader.ckSize = sizeof (WaveHeader);
+
+ WaveHeader.FormatTag = 1;
+ WaveHeader.NumChannels = num_channels;
+ WaveHeader.SampleRate = WavpackGetSampleRate (wpc);
+ WaveHeader.BlockAlign = num_channels * bps;
+ WaveHeader.BytesPerSecond = WaveHeader.SampleRate * WaveHeader.BlockAlign;
+ WaveHeader.BitsPerSample = WavpackGetBitsPerSample (wpc);
+
+ strncpy (DataChunkHeader.ckID, "data", sizeof (DataChunkHeader.ckID));
+ DataChunkHeader.ckSize = total_samples * num_channels * bps;
+
+ native_to_little_endian (&RiffChunkHeader, ChunkHeaderFormat);
+ native_to_little_endian (&FormatChunkHeader, ChunkHeaderFormat);
+ native_to_little_endian (&WaveHeader, WaveHeaderFormat);
+ native_to_little_endian (&DataChunkHeader, ChunkHeaderFormat);
+
+ if (!fwrite (&RiffChunkHeader, sizeof (RiffChunkHeader), 1, stdout) ||
+ !fwrite (&FormatChunkHeader, sizeof (FormatChunkHeader), 1, stdout) ||
+ !fwrite (&WaveHeader, sizeof (WaveHeader), 1, stdout) ||
+ !fwrite (&DataChunkHeader, sizeof (DataChunkHeader), 1, stdout)) {
+ fputs ("can't write .WAV data, disk probably full!\n", stderr);
+ return 1;
+ }
+
+ while (1) {
+ uint32_t samples_unpacked;
+
+ samples_unpacked = WavpackUnpackSamples (wpc, temp_buffer, 256 / num_channels);
+ total_unpacked_samples += samples_unpacked;
+
+ if (samples_unpacked) {
+ format_samples (bps, (uchar *) temp_buffer, temp_buffer, samples_unpacked *= num_channels);
+
+ if (fwrite (temp_buffer, bps, samples_unpacked, stdout) != samples_unpacked) {
+ fputs ("can't write .WAV data, disk probably full!\n", stderr);
+ return 1;
+ }
+ }
+
+ if (!samples_unpacked)
+ break;
+ }
+
+ fflush (stdout);
+
+ if (WavpackGetNumSamples (wpc) != (uint32_t) -1 &&
+ total_unpacked_samples != WavpackGetNumSamples (wpc)) {
+ fputs ("incorrect number of samples!\n", stderr);
+ return 1;
+ }
+
+ if (WavpackGetNumErrors (wpc)) {
+ fputs ("crc errors detected!\n", stderr);
+ return 1;
+ }
+
+ return 0;
+}
+
+static int32_t read_bytes (void *buff, int32_t bcount)
+{
+ return fread (buff, 1, bcount, stdin);
+}
+
+// Reformat samples from longs in processor's native endian mode to
+// little-endian data with (possibly) less than 4 bytes / sample.
+
+static uchar *format_samples (int bps, uchar *dst, int32_t *src, uint32_t samcnt)
+{
+ int32_t temp;
+
+ switch (bps) {
+
+ case 1:
+ while (samcnt--)
+ *dst++ = *src++ + 128;
+
+ break;
+
+ case 2:
+ while (samcnt--) {
+ *dst++ = (uchar)(temp = *src++);
+ *dst++ = (uchar)(temp >> 8);
+ }
+
+ break;
+
+ case 3:
+ while (samcnt--) {
+ *dst++ = (uchar)(temp = *src++);
+ *dst++ = (uchar)(temp >> 8);
+ *dst++ = (uchar)(temp >> 16);
+ }
+
+ break;
+
+ case 4:
+ while (samcnt--) {
+ *dst++ = (uchar)(temp = *src++);
+ *dst++ = (uchar)(temp >> 8);
+ *dst++ = (uchar)(temp >> 16);
+ *dst++ = (uchar)(temp >> 24);
+ }
+
+ break;
+ }
+
+ return dst;
+}
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