apps/codecs/libspeex/lsp.h


1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64

/*---------------------------------------------------------------------------*\
Original Copyright
    FILE........: AK2LSPD.H
    TYPE........: Turbo C header file
    COMPANY.....: Voicetronix
    AUTHOR......: James Whitehall
    DATE CREATED: 21/11/95

Modified by Jean-Marc Valin

    This file contains functions for converting Linear Prediction
    Coefficients (LPC) to Line Spectral Pair (LSP) and back. Note that the
    LSP coefficients are not in radians format but in the x domain of the
    unit circle.

\*---------------------------------------------------------------------------*/
/**
   @file lsp.h
   @brief Line Spectral Pair (LSP) functions.
*/
/* Speex License:

   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 the Xiph.org Foundation 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 FOUNDATION 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.
*/

#ifndef __AK2LSPD__
#define __AK2LSPD__

#include "arch.h"

int lpc_to_lsp (spx_coef_t *a, int lpcrdr, spx_lsp_t *freq, int nb, spx_word16_t delta, char *stack);
void lsp_to_lpc(spx_lsp_t *freq, spx_coef_t *ak, int lpcrdr, char *stack);

/*Added by JMV*/
void lsp_enforce_margin(spx_lsp_t *lsp, int len, spx_word16_t margin);

void lsp_interpolate(spx_lsp_t *old_lsp, spx_lsp_t *new_lsp, spx_lsp_t *interp_lsp, int len, int subframe, int nb_subframes);

#endif  /* __AK2LSPD__ */
////////////////////////////////////////////////////////////////////////////
//                           **** WAVPACK ****                            //
//                  Hybrid Lossless Wavefile Compressor                   //
//              Copyright (c) 1998 - 2004 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>

static void strcpy_loc (char *dst, char *src) { while ((*dst++ = *src++) != 0); }

///////////////////////////// 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, 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).

static WavpackContext wpc IBSS_ATTR;

WavpackContext *WavpackOpenFileInput (read_stream infile, char *error)
{
    WavpackStream *wps = &wpc.stream;
    uint32_t bcount;

    CLEAR (wpc);
    wpc.infile = infile;
    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, &wps->wphdr);

        if (bcount == (uint32_t) -1) {
            strcpy_loc (error, "invalid WavPack file!");
            return NULL;
        }

        if ((wps->wphdr.flags & UNKNOWN_FLAGS) || wps->wphdr.version < MIN_STREAM_VERS ||
            wps->wphdr.version > MAX_STREAM_VERS) {
                strcpy_loc (error, "invalid WavPack file!");
                return NULL;
        }

        if (wps->wphdr.block_samples && wps->wphdr.total_samples != (uint32_t) -1)
            wpc.total_samples = wps->wphdr.total_samples;

        if (!unpack_init (&wpc)) {
            strcpy_loc (error, wpc.error_message [0] ? wpc.error_message :
                "invalid WavPack file!");

            return NULL;
        }
    }

    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.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 &wpc;
}

// 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
// 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 int32_t's for stereo files. The audio data is returned right-justified in
// 32-bit int32_t's 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 can also be returned if the source was
// floating point data (and this is normalized to +/-1.0). 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, &wps->wphdr);

                if (bcount == (uint32_t) -1)
                    break;

                if (wps->wphdr.version < MIN_STREAM_VERS || wps->wphdr.version > MAX_STREAM_VERS) {
                    strcpy_loc (wpc->error_message, "invalid WavPack file!");
                    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 int32_t's, 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 int32_t buffer and must be
// left-shifted 8, 16, or 24 bits if normalized int32_t's 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, 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 (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) > 1024 * 1024)
            return -1;
    }
}

// Open context for writing WavPack files. The returned context pointer is used
// in all following calls to the library. A return value of NULL indicates
// that memory could not be allocated for the context.

WavpackContext *WavpackOpenFileOutput (void)
{
    CLEAR (wpc);
    return &wpc;
}

// 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->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

// config->bitrate              hybrid bitrate in either bits/sample or kbps
// config->shaping_weight       hybrid noise shaping coefficient override
// config->float_norm_exp       select floating-point data (127 for +/-1.0)

// 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.

int WavpackSetConfiguration (WavpackContext *wpc, WavpackConfig *config, uint32_t total_samples)
{
    WavpackStream *wps = &wpc->stream;
    uint32_t flags = (config->bytes_per_sample - 1), shift = 0;
    int num_chans = config->num_channels;
    int i;

    if ((wpc->config.flags & CONFIG_HYBRID_FLAG) ||
        wpc->config.float_norm_exp ||
        num_chans < 1 || num_chans > 2)
            return FALSE;

    wpc->total_samples = total_samples;
    wpc->config.sample_rate = config->sample_rate;
    wpc->config.num_channels = config->num_channels;
    wpc->config.bits_per_sample = config->bits_per_sample;
    wpc->config.bytes_per_sample = config->bytes_per_sample;
    wpc->config.flags = config->flags;

    shift = (config->bytes_per_sample * 8) - config->bits_per_sample;

    for (i = 0; i < 15; ++i)
        if (wpc->config.sample_rate == sample_rates [i])
            break;

    flags |= i << SRATE_LSB;
    flags |= shift << SHIFT_LSB;
    flags |= CROSS_DECORR;

    if (!(config->flags & CONFIG_JOINT_OVERRIDE) || (config->flags & CONFIG_JOINT_STEREO))
        flags |= JOINT_STEREO;

    flags |= INITIAL_BLOCK | FINAL_BLOCK;

    if (num_chans == 1) {
        flags &= ~(JOINT_STEREO | CROSS_DECORR | HYBRID_BALANCE);
        flags |= MONO_FLAG;
    }

    flags &= ~MAG_MASK;
    flags += (1 << MAG_LSB) * ((flags & BYTES_STORED) * 8 + 7);

    memcpy (wps->wphdr.ckID, "wvpk", 4);
    wps->wphdr.ckSize = sizeof (WavpackHeader) - 8;
    wps->wphdr.total_samples = wpc->total_samples;
    wps->wphdr.version = CUR_STREAM_VERS;
    wps->wphdr.flags = flags;

    pack_init (wpc);
    return TRUE;
}

// Add wrapper (currently RIFF only) to WavPack blocks. This should be called
// before sending any audio samples. 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.

void WavpackAddWrapper (WavpackContext *wpc, void *data, uint32_t bcount)
{
    wpc->wrapper_data = data;
    wpc->wrapper_bytes = bcount;
}

// Start a WavPack block to be stored in the specified buffer. This must be
// called before calling WavpackPackSamples(). Note that writing CANNOT wrap
// in the buffer; the entire output block must fit in the buffer.

int WavpackStartBlock (WavpackContext *wpc, uchar *begin, uchar *end)
{
    wpc->stream.blockbuff = begin;
    wpc->stream.blockend = end;
    return pack_start_block (wpc);
}

// Pack the specified samples. Samples must be stored in int32_ts 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. The caller must decide how many samples to place in each
// WavPack block (1/2 second is common), but this function may be called as
// many times as desired to build the final block (and performs the actual
// compression during the call). A return of FALSE indicates an error.

int WavpackPackSamples (WavpackContext *wpc, int32_t *sample_buffer, uint32_t sample_count)
{
    if (!sample_count || pack_samples (wpc, sample_buffer, sample_count))
        return TRUE;

    strcpy_loc (wpc->error_message, "output buffer overflowed!");
    return FALSE;
}

// Finish the WavPack block being built, returning the total size of the
// block in bytes. Note that the possible conversion of the WavPack header to
// little-endian takes place here.

uint32_t WavpackFinishBlock (WavpackContext *wpc)
{
    WavpackStream *wps = &wpc->stream;
    uint32_t bcount;

    pack_finish_block (wpc);
    bcount = ((WavpackHeader *) wps->blockbuff)->ckSize + 8;
    native_to_little_endian ((WavpackHeader *) wps->blockbuff, WavpackHeaderFormat);

    return bcount;
}

// 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. 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.

void WavpackUpdateNumSamples (WavpackContext *wpc, void *first_block)
{
    little_endian_to_native (wpc, WavpackHeaderFormat);
    ((WavpackHeader *) first_block)->total_samples = WavpackGetSampleIndex (wpc);
    native_to_little_endian (wpc, WavpackHeaderFormat);
}

// 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.
// It is the responsibility of the application to read and rewrite the block.
// An example of this can be found in the Audition filter.

void *WavpackGetWrapperLocation (void *first_block)
{
    if (((uchar *) first_block) [32] == ID_RIFF_HEADER)
        return ((uchar *) first_block) + 34;
    else
        return NULL;
}