ref: 8f6aaf97498ff7edc0dcb30192542b4d5924411c
dir: /src/wav.c/
/* libSoX microsoft's WAVE sound format handler
*
* Copyright 1998-2006 Chris Bagwell and SoX Contributors
* Copyright 1997 Graeme W. Gill, 93/5/17
* Copyright 1992 Rick Richardson
* Copyright 1991 Lance Norskog And Sundry Contributors
*
* Info for format tags can be found at:
* http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
*
*/
#include "sox_i.h"
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include "ima_rw.h"
#include "adpcm.h"
#ifdef HAVE_LIBGSM
#ifdef HAVE_GSM_GSM_H
#include <gsm/gsm.h>
#else
#include <gsm.h>
#endif
#endif
/* Magic length sometimes used to indicate unknown or too large size.
* When detected on inputs, disable any length logic.
*/
#define MS_UNSPEC 0x7ffff000
#define WAVE_FORMAT_UNKNOWN 0x0000
#define WAVE_FORMAT_PCM 0x0001
#define WAVE_FORMAT_ADPCM 0x0002
#define WAVE_FORMAT_IEEE_FLOAT 0x0003
#define WAVE_FORMAT_IBM_CVSD 0x0005
#define WAVE_FORMAT_ALAW 0x0006
#define WAVE_FORMAT_MULAW 0x0007
#define WAVE_FORMAT_OKI_ADPCM 0x0010
#define WAVE_FORMAT_IMA_ADPCM 0x0011
#define WAVE_FORMAT_MEDIASPACE_ADPCM 0x0012
#define WAVE_FORMAT_SIERRA_ADPCM 0x0013
#define WAVE_FORMAT_G723_ADPCM 0x0014
#define WAVE_FORMAT_DIGISTD 0x0015
#define WAVE_FORMAT_DIGIFIX 0x0016
#define WAVE_FORMAT_YAMAHA_ADPCM 0x0020
#define WAVE_FORMAT_SONARC 0x0021
#define WAVE_FORMAT_TRUESPEECH 0x0022
#define WAVE_FORMAT_ECHOSC1 0x0023
#define WAVE_FORMAT_AUDIOFILE_AF36 0x0024
#define WAVE_FORMAT_APTX 0x0025
#define WAVE_FORMAT_AUDIOFILE_AF10 0x0026
#define WAVE_FORMAT_DOLBY_AC2 0x0030
#define WAVE_FORMAT_GSM610 0x0031
#define WAVE_FORMAT_ADPCME 0x0033
#define WAVE_FORMAT_CONTROL_RES_VQLPC 0x0034
#define WAVE_FORMAT_DIGIREAL 0x0035
#define WAVE_FORMAT_DIGIADPCM 0x0036
#define WAVE_FORMAT_CONTROL_RES_CR10 0x0037
#define WAVE_FORMAT_ROCKWELL_ADPCM 0x003b
#define WAVE_FORMAT_ROCKWELL_DIGITALK 0x003c
#define WAVE_FORMAT_G721_ADPCM 0x0040
#define WAVE_FORMAT_G728_CELP 0x0041
#define WAVE_FORMAT_MPEG 0x0050
#define WAVE_FORMAT_MPEGLAYER3 0x0055
#define WAVE_FORMAT_G726_ADPCM 0x0064
#define WAVE_FORMAT_G722_ADPCM 0x0065
#define WAVE_FORMAT_CREATIVE_ADPCM 0x0200
#define WAVE_FORMAT_CREATIVE_FSP8 0x0202
#define WAVE_FORMAT_CREATIVE_FSP10 0x0203
#define WAVE_FORMAT_FM_TOWNS_SND 0x0300
#define WAVE_FORMAT_OLIGSM 0x1000
#define WAVE_FORMAT_OLIADPCM 0x1001
#define WAVE_FORMAT_OLISBC 0x1003
#define WAVE_FORMAT_OLIOPR 0x1004
#define WAVE_FORMAT_EXTENSIBLE 0xfffe
/* To allow padding to samplesPerBlock. Works, but currently never true. */
static const size_t pad_nsamps = sox_false;
/* Private data for .wav file */
typedef struct {
/* samples/channel reading: starts at total count and decremented */
/* writing: starts at 0 and counts samples written */
uint64_t numSamples;
size_t dataLength; /* needed for ADPCM writing */
unsigned short formatTag; /* What type of encoding file is using */
unsigned short samplesPerBlock;
unsigned short blockAlign;
uint16_t bitsPerSample; /* bits per sample */
size_t dataStart; /* need to for seeking */
int ignoreSize; /* ignoreSize allows us to process 32-bit WAV files that are
* greater then 2 Gb and can't be represented by the
* 32-bit size field. */
/* FIXME: Have some front-end code which sets this flag. */
/* following used by *ADPCM wav files */
unsigned short nCoefs; /* ADPCM: number of coef sets */
short *lsx_ms_adpcm_i_coefs; /* ADPCM: coef sets */
void *ms_adpcm_data; /* Private data of adpcm decoder */
unsigned char *packet; /* Temporary buffer for packets */
short *samples; /* interleaved samples buffer */
short *samplePtr; /* Pointer to current sample */
short *sampleTop; /* End of samples-buffer */
unsigned short blockSamplesRemaining;/* Samples remaining per channel */
int state[16]; /* step-size info for *ADPCM writes */
#ifdef HAVE_LIBGSM
/* following used by GSM 6.10 wav */
gsm gsmhandle;
gsm_signal *gsmsample;
int gsmindex;
size_t gsmbytecount; /* counts bytes written to data block */
#endif
} priv_t;
struct wave_format {
uint16_t tag;
const char *name;
sox_encoding_t encoding;
int (*read_fmt)(sox_format_t *ft, uint32_t len);
};
static const char *wav_format_str(unsigned tag);
static int wavwritehdr(sox_format_t *, int);
/****************************************************************************/
/* IMA ADPCM Support Functions Section */
/****************************************************************************/
static int wav_ima_adpcm_fmt(sox_format_t *ft, uint32_t len)
{
priv_t *wav = ft->priv;
size_t bytesPerBlock;
int err;
if (wav->bitsPerSample != 4) {
lsx_fail_errno(ft, SOX_EOF,
"Can only handle 4-bit IMA ADPCM in wav files");
return SOX_EOF;
}
err = lsx_read_fields(ft, &len, "h", &wav->samplesPerBlock);
if (err)
return SOX_EOF;
bytesPerBlock = lsx_ima_bytes_per_block(ft->signal.channels,
wav->samplesPerBlock);
if (bytesPerBlock != wav->blockAlign || wav->samplesPerBlock % 8 != 1) {
lsx_fail_errno(ft, SOX_EOF,
"format[%s]: samplesPerBlock(%d) != blockAlign(%d)",
wav_format_str(wav->formatTag),
wav->samplesPerBlock, wav->blockAlign);
return SOX_EOF;
}
wav->packet = lsx_malloc(wav->blockAlign);
wav->samples =
lsx_malloc(ft->signal.channels * wav->samplesPerBlock * sizeof(short));
return SOX_SUCCESS;
}
/*
*
* ImaAdpcmReadBlock - Grab and decode complete block of samples
*
*/
static unsigned short ImaAdpcmReadBlock(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
size_t bytesRead;
int samplesThisBlock;
/* Pull in the packet and check the header */
bytesRead = lsx_readbuf(ft, wav->packet, (size_t)wav->blockAlign);
samplesThisBlock = wav->samplesPerBlock;
if (bytesRead < wav->blockAlign)
{
/* If it looks like a valid header is around then try and */
/* work with partial blocks. Specs say it should be null */
/* padded but I guess this is better than trailing quiet. */
samplesThisBlock = lsx_ima_samples_in((size_t)0, (size_t)ft->signal.channels, bytesRead, (size_t) 0);
if (samplesThisBlock == 0 || samplesThisBlock > wav->samplesPerBlock)
{
lsx_warn("Premature EOF on .wav input file");
return 0;
}
}
wav->samplePtr = wav->samples;
/* For a full block, the following should be true: */
/* wav->samplesPerBlock = blockAlign - 8byte header + 1 sample in header */
lsx_ima_block_expand_i(ft->signal.channels, wav->packet, wav->samples, samplesThisBlock);
return samplesThisBlock;
}
/****************************************************************************/
/* MS ADPCM Support Functions Section */
/****************************************************************************/
static int wav_ms_adpcm_fmt(sox_format_t *ft, uint32_t len)
{
priv_t *wav = ft->priv;
size_t bytesPerBlock;
int i, errct = 0;
int err;
if (wav->bitsPerSample != 4) {
lsx_fail_errno(ft, SOX_EOF,
"Can only handle 4-bit MS ADPCM in wav files");
return SOX_EOF;
}
err = lsx_read_fields(ft, &len, "hh", &wav->samplesPerBlock, &wav->nCoefs);
if (err)
return SOX_EOF;
bytesPerBlock = lsx_ms_adpcm_bytes_per_block(ft->signal.channels,
wav->samplesPerBlock);
if (bytesPerBlock != wav->blockAlign) {
lsx_fail_errno(ft, SOX_EOF,
"format[%s]: samplesPerBlock(%d) != blockAlign(%d)",
wav_format_str(wav->formatTag),
wav->samplesPerBlock, wav->blockAlign);
return SOX_EOF;
}
if (wav->nCoefs < 7 || wav->nCoefs > 0x100) {
lsx_fail_errno(ft, SOX_EOF,
"ADPCM file nCoefs (%.4hx) makes no sense",
wav->nCoefs);
return SOX_EOF;
}
if (len < 4 * wav->nCoefs) {
lsx_fail_errno(ft, SOX_EOF, "wave header error: cbSize too small");
return SOX_EOF;
}
wav->packet = lsx_malloc(wav->blockAlign);
wav->samples =
lsx_malloc(ft->signal.channels * wav->samplesPerBlock * sizeof(short));
/* nCoefs, lsx_ms_adpcm_i_coefs used by adpcm.c */
wav->lsx_ms_adpcm_i_coefs = lsx_malloc(wav->nCoefs * 2 * sizeof(short));
wav->ms_adpcm_data = lsx_ms_adpcm_alloc(ft->signal.channels);
err = lsx_read_fields(ft, &len, "*h",
2 * wav->nCoefs, wav->lsx_ms_adpcm_i_coefs);
if (err)
return SOX_EOF;
for (i = 0; i < 14; i++)
errct += wav->lsx_ms_adpcm_i_coefs[i] != lsx_ms_adpcm_i_coef[i/2][i%2];
if (errct)
lsx_warn("base lsx_ms_adpcm_i_coefs differ in %d/14 positions", errct);
return SOX_SUCCESS;
}
/*
*
* AdpcmReadBlock - Grab and decode complete block of samples
*
*/
static unsigned short AdpcmReadBlock(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
size_t bytesRead;
int samplesThisBlock;
const char *errmsg;
/* Pull in the packet and check the header */
bytesRead = lsx_readbuf(ft, wav->packet, (size_t) wav->blockAlign);
samplesThisBlock = wav->samplesPerBlock;
if (bytesRead < wav->blockAlign)
{
/* If it looks like a valid header is around then try and */
/* work with partial blocks. Specs say it should be null */
/* padded but I guess this is better than trailing quiet. */
samplesThisBlock = lsx_ms_adpcm_samples_in((size_t)0, (size_t)ft->signal.channels, bytesRead, (size_t)0);
if (samplesThisBlock == 0 || samplesThisBlock > wav->samplesPerBlock)
{
lsx_warn("Premature EOF on .wav input file");
return 0;
}
}
errmsg = lsx_ms_adpcm_block_expand_i(wav->ms_adpcm_data, ft->signal.channels, wav->nCoefs, wav->lsx_ms_adpcm_i_coefs, wav->packet, wav->samples, samplesThisBlock);
if (errmsg)
lsx_warn("%s", errmsg);
return samplesThisBlock;
}
/****************************************************************************/
/* Common ADPCM Write Function */
/****************************************************************************/
static int xxxAdpcmWriteBlock(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
size_t chans, ct;
short *p;
chans = ft->signal.channels;
p = wav->samplePtr;
ct = p - wav->samples;
if (ct>=chans) {
/* zero-fill samples if needed to complete block */
for (p = wav->samplePtr; p < wav->sampleTop; p++) *p=0;
/* compress the samples to wav->packet */
if (wav->formatTag == WAVE_FORMAT_ADPCM) {
lsx_ms_adpcm_block_mash_i((unsigned) chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, wav->blockAlign);
}else{ /* WAVE_FORMAT_IMA_ADPCM */
lsx_ima_block_mash_i((unsigned) chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, 9);
}
/* write the compressed packet */
if (lsx_writebuf(ft, wav->packet, (size_t) wav->blockAlign) != wav->blockAlign)
{
lsx_fail_errno(ft,SOX_EOF,"write error");
return (SOX_EOF);
}
/* update lengths and samplePtr */
wav->dataLength += wav->blockAlign;
if (pad_nsamps)
wav->numSamples += wav->samplesPerBlock;
else
wav->numSamples += ct/chans;
wav->samplePtr = wav->samples;
}
return (SOX_SUCCESS);
}
#ifdef HAVE_LIBGSM
/****************************************************************************/
/* WAV GSM6.10 support functions */
/****************************************************************************/
static int wav_gsm_fmt(sox_format_t *ft, uint32_t len)
{
priv_t *wav = ft->priv;
int err;
err = lsx_read_fields(ft, &len, "h", &wav->samplesPerBlock);
if (err)
return SOX_EOF;
if (wav->blockAlign != 65) {
lsx_fail_errno(ft, SOX_EOF, "format[%s]: expects blockAlign(%d) = %d",
wav_format_str(wav->formatTag), wav->blockAlign, 65);
return SOX_EOF;
}
if (wav->samplesPerBlock != 320) {
lsx_fail_errno(ft, SOX_EOF,
"format[%s]: expects samplesPerBlock(%d) = %d",
wav_format_str(wav->formatTag),
wav->samplesPerBlock, 320);
return SOX_EOF;
}
return SOX_SUCCESS;
}
/* create the gsm object, malloc buffer for 160*2 samples */
static int wavgsminit(sox_format_t * ft)
{
int valueP=1;
priv_t * wav = (priv_t *) ft->priv;
wav->gsmbytecount=0;
wav->gsmhandle=gsm_create();
if (!wav->gsmhandle)
{
lsx_fail_errno(ft,SOX_EOF,"cannot create GSM object");
return (SOX_EOF);
}
if(gsm_option(wav->gsmhandle,GSM_OPT_WAV49,&valueP) == -1){
lsx_fail_errno(ft,SOX_EOF,"error setting gsm_option for WAV49 format. Recompile gsm library with -DWAV49 option and relink sox");
return (SOX_EOF);
}
wav->gsmsample=lsx_malloc(sizeof(gsm_signal)*160*2);
wav->gsmindex=0;
return (SOX_SUCCESS);
}
/*destroy the gsm object and free the buffer */
static void wavgsmdestroy(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
gsm_destroy(wav->gsmhandle);
free(wav->gsmsample);
}
static size_t wavgsmread(sox_format_t * ft, sox_sample_t *buf, size_t len)
{
priv_t * wav = (priv_t *) ft->priv;
size_t done=0;
int bytes;
gsm_byte frame[65];
ft->sox_errno = SOX_SUCCESS;
/* copy out any samples left from the last call */
while(wav->gsmindex && (wav->gsmindex<160*2) && (done < len))
buf[done++]=SOX_SIGNED_16BIT_TO_SAMPLE(wav->gsmsample[wav->gsmindex++],);
/* read and decode loop, possibly leaving some samples in wav->gsmsample */
while (done < len) {
wav->gsmindex=0;
bytes = lsx_readbuf(ft, frame, (size_t)65);
if (bytes <=0)
return done;
if (bytes<65) {
lsx_warn("invalid wav gsm frame size: %d bytes",bytes);
return done;
}
/* decode the long 33 byte half */
if(gsm_decode(wav->gsmhandle,frame, wav->gsmsample)<0)
{
lsx_fail_errno(ft,SOX_EOF,"error during gsm decode");
return 0;
}
/* decode the short 32 byte half */
if(gsm_decode(wav->gsmhandle,frame+33, wav->gsmsample+160)<0)
{
lsx_fail_errno(ft,SOX_EOF,"error during gsm decode");
return 0;
}
while ((wav->gsmindex <160*2) && (done < len)){
buf[done++]=SOX_SIGNED_16BIT_TO_SAMPLE(wav->gsmsample[(wav->gsmindex)++],);
}
}
return done;
}
static int wavgsmflush(sox_format_t * ft)
{
gsm_byte frame[65];
priv_t * wav = (priv_t *) ft->priv;
/* zero fill as needed */
while(wav->gsmindex<160*2)
wav->gsmsample[wav->gsmindex++]=0;
/*encode the even half short (32 byte) frame */
gsm_encode(wav->gsmhandle, wav->gsmsample, frame);
/*encode the odd half long (33 byte) frame */
gsm_encode(wav->gsmhandle, wav->gsmsample+160, frame+32);
if (lsx_writebuf(ft, frame, (size_t) 65) != 65)
{
lsx_fail_errno(ft,SOX_EOF,"write error");
return (SOX_EOF);
}
wav->gsmbytecount += 65;
wav->gsmindex = 0;
return (SOX_SUCCESS);
}
static size_t wavgsmwrite(sox_format_t * ft, const sox_sample_t *buf, size_t len)
{
priv_t * wav = (priv_t *) ft->priv;
size_t done = 0;
int rc;
ft->sox_errno = SOX_SUCCESS;
while (done < len) {
SOX_SAMPLE_LOCALS;
while ((wav->gsmindex < 160*2) && (done < len))
wav->gsmsample[(wav->gsmindex)++] =
SOX_SAMPLE_TO_SIGNED_16BIT(buf[done++], ft->clips);
if (wav->gsmindex < 160*2)
break;
rc = wavgsmflush(ft);
if (rc)
return 0;
}
return done;
}
static void wavgsmstopwrite(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
ft->sox_errno = SOX_SUCCESS;
if (wav->gsmindex)
wavgsmflush(ft);
/* Add a pad byte if amount of written bytes is not even. */
if (wav->gsmbytecount && wav->gsmbytecount % 2){
if(lsx_writeb(ft, 0))
lsx_fail_errno(ft,SOX_EOF,"write error");
else
wav->gsmbytecount += 1;
}
wavgsmdestroy(ft);
}
#endif /* HAVE_LIBGSM */
/****************************************************************************/
/* General Sox WAV file code */
/****************************************************************************/
static int wav_pcm_fmt(sox_format_t *ft, uint32_t len)
{
priv_t *wav = ft->priv;
int bps = (wav->bitsPerSample + 7) / 8;
if (bps == 1) {
ft->encoding.encoding = SOX_ENCODING_UNSIGNED;
} else if (bps <= 4) {
ft->encoding.encoding = SOX_ENCODING_SIGN2;
} else {
lsx_fail_errno(ft, SOX_EFMT, "%d bytes per sample not suppored", bps);
return SOX_EOF;
}
return SOX_SUCCESS;
}
static const struct wave_format wave_formats[] = {
{ WAVE_FORMAT_UNKNOWN, "Unknown Wave Type" },
{ WAVE_FORMAT_PCM, "PCM",
SOX_ENCODING_UNKNOWN,
wav_pcm_fmt,
},
{ WAVE_FORMAT_ADPCM, "Microsoft ADPCM",
SOX_ENCODING_MS_ADPCM,
wav_ms_adpcm_fmt,
},
{ WAVE_FORMAT_IEEE_FLOAT, "IEEE Float",
SOX_ENCODING_FLOAT },
{ WAVE_FORMAT_IBM_CVSD, "Digispeech CVSD" },
{ WAVE_FORMAT_ALAW, "CCITT A-law",
SOX_ENCODING_ALAW },
{ WAVE_FORMAT_MULAW, "CCITT u-law",
SOX_ENCODING_ULAW },
{ WAVE_FORMAT_OKI_ADPCM, "OKI ADPCM" },
{ WAVE_FORMAT_IMA_ADPCM, "IMA ADPCM",
SOX_ENCODING_IMA_ADPCM,
wav_ima_adpcm_fmt,
},
{ WAVE_FORMAT_MEDIASPACE_ADPCM, "MediaSpace ADPCM" },
{ WAVE_FORMAT_SIERRA_ADPCM, "Sierra ADPCM" },
{ WAVE_FORMAT_G723_ADPCM, "G.723 ADPCM" },
{ WAVE_FORMAT_DIGISTD, "DIGISTD" },
{ WAVE_FORMAT_DIGIFIX, "DigiFix" },
{ WAVE_FORMAT_YAMAHA_ADPCM, "Yamaha ADPCM" },
{ WAVE_FORMAT_SONARC, "Sonarc" },
{ WAVE_FORMAT_TRUESPEECH, "Truespeech" },
{ WAVE_FORMAT_ECHOSC1, "ECHO SC-1", },
{ WAVE_FORMAT_AUDIOFILE_AF36, "Audio File AF36" },
{ WAVE_FORMAT_APTX, "aptX" },
{ WAVE_FORMAT_AUDIOFILE_AF10, "Audio File AF10" },
{ WAVE_FORMAT_DOLBY_AC2, "Dolby AC-2" },
{ WAVE_FORMAT_GSM610, "GSM 6.10",
#ifdef HAVE_LIBGSM
SOX_ENCODING_GSM,
wav_gsm_fmt,
#endif
},
{ WAVE_FORMAT_ADPCME, "Antex ADPCME" },
{ WAVE_FORMAT_CONTROL_RES_VQLPC, "Control Resources VQLPC" },
{ WAVE_FORMAT_DIGIREAL, "DSP Solutions REAL" },
{ WAVE_FORMAT_DIGIADPCM, "DSP Solutions ADPCM" },
{ WAVE_FORMAT_CONTROL_RES_CR10, "Control Resources CR10" },
{ WAVE_FORMAT_ROCKWELL_ADPCM, "Rockwell ADPCM" },
{ WAVE_FORMAT_ROCKWELL_DIGITALK, "Rockwell DIGITALK" },
{ WAVE_FORMAT_G721_ADPCM, "G.721 ADPCM" },
{ WAVE_FORMAT_G728_CELP, "G.728 CELP" },
{ WAVE_FORMAT_MPEG, "MPEG-1 Audio" },
{ WAVE_FORMAT_MPEGLAYER3, "MPEG-1 Layer 3" },
{ WAVE_FORMAT_G726_ADPCM, "G.726 ADPCM" },
{ WAVE_FORMAT_G722_ADPCM, "G.722 ADPCM" },
{ WAVE_FORMAT_CREATIVE_ADPCM, "Creative Labs ADPCM" },
{ WAVE_FORMAT_CREATIVE_FSP8, "Creative Labs FastSpeech 8" },
{ WAVE_FORMAT_CREATIVE_FSP10, "Creative Labs FastSpeech 10" },
{ WAVE_FORMAT_FM_TOWNS_SND, "Fujitsu FM Towns SND" },
{ WAVE_FORMAT_OLIGSM, "Olivetti GSM" },
{ WAVE_FORMAT_OLIADPCM, "Olivetti ADPCM" },
{ WAVE_FORMAT_OLISBC, "Olivetti CELP" },
{ WAVE_FORMAT_OLIOPR, "Olivetti OPR" },
{ }
};
static const struct wave_format *wav_find_format(unsigned tag)
{
const struct wave_format *f;
for (f = wave_formats; f->name; f++)
if (f->tag == tag)
return f;
return NULL;
}
static int wavfail(sox_format_t *ft, int tag, const char *name)
{
if (name)
lsx_fail_errno(ft, SOX_EHDR, "WAVE format '%s' (%04x) not supported",
name, tag);
else
lsx_fail_errno(ft, SOX_EHDR, "Unknown WAVE format %04x", tag);
return SOX_EOF;
}
static int wav_read_fmt(sox_format_t *ft, uint32_t len)
{
priv_t *wav = ft->priv;
uint16_t wChannels; /* number of channels */
uint32_t dwSamplesPerSecond; /* samples per second per channel */
uint32_t dwAvgBytesPerSec; /* estimate of bytes per second needed */
uint16_t wExtSize = 0; /* extended field for non-PCM */
const struct wave_format *fmt;
sox_encoding_t user_enc = ft->encoding.encoding;
int err;
if (len < 16) {
lsx_fail_errno(ft, SOX_EHDR, "WAVE file fmt chunk is too short");
return SOX_EOF;
}
err = lsx_read_fields(ft, &len, "hhiihh",
&wav->formatTag,
&wChannels,
&dwSamplesPerSecond,
&dwAvgBytesPerSec,
&wav->blockAlign,
&wav->bitsPerSample);
if (err)
return SOX_EOF;
/* non-PCM formats except alaw and mulaw formats have extended fmt chunk.
* Check for those cases.
*/
if (wav->formatTag != WAVE_FORMAT_PCM &&
wav->formatTag != WAVE_FORMAT_ALAW &&
wav->formatTag != WAVE_FORMAT_MULAW &&
len < 2)
lsx_warn("WAVE file missing extended part of fmt chunk");
if (len >= 2) {
err = lsx_read_fields(ft, &len, "h", &wExtSize);
if (err)
return SOX_EOF;
}
if (wExtSize != len) {
lsx_fail_errno(ft, SOX_EOF,
"WAVE header error: cbSize inconsistent with fmt size");
return SOX_EOF;
}
if (wav->formatTag == WAVE_FORMAT_EXTENSIBLE) {
uint16_t numberOfValidBits;
uint32_t speakerPositionMask;
uint16_t subFormatTag;
if (len < 22) {
lsx_fail_errno(ft, SOX_EHDR, "WAVE file fmt chunk is too short");
return SOX_EOF;
}
err = lsx_read_fields(ft, &len, "hih14x",
&numberOfValidBits,
&speakerPositionMask,
&subFormatTag);
if (err)
return SOX_EOF;
if (numberOfValidBits > wav->bitsPerSample) {
lsx_fail_errno(ft, SOX_EHDR,
"wValidBitsPerSample > wBitsPerSample");
return SOX_EOF;
}
wav->formatTag = subFormatTag;
lsx_report("EXTENSIBLE");
}
/* User options take precedence */
if (ft->signal.channels == 0 || ft->signal.channels == wChannels)
ft->signal.channels = wChannels;
else
lsx_report("User options overriding channels read in .wav header");
if (ft->signal.channels == 0) {
lsx_fail_errno(ft, SOX_EHDR, "Channel count is zero");
return SOX_EOF;
}
if (ft->signal.rate == 0 || ft->signal.rate == dwSamplesPerSecond)
ft->signal.rate = dwSamplesPerSecond;
else
lsx_report("User options overriding rate read in .wav header");
fmt = wav_find_format(wav->formatTag);
if (!fmt)
return wavfail(ft, wav->formatTag, NULL);
/* format handler might override */
ft->encoding.encoding = fmt->encoding;
if (fmt->read_fmt) {
if (fmt->read_fmt(ft, len))
return SOX_EOF;
} else if (!fmt->encoding) {
return wavfail(ft, wav->formatTag, fmt->name);
}
/* User options take precedence */
if (!ft->encoding.bits_per_sample ||
ft->encoding.bits_per_sample == wav->bitsPerSample)
ft->encoding.bits_per_sample = wav->bitsPerSample;
else
lsx_warn("User options overriding size read in .wav header");
if (user_enc && user_enc != ft->encoding.encoding) {
lsx_report("User options overriding encoding read in .wav header");
ft->encoding.encoding = user_enc;
}
return 0;
}
static sox_bool valid_chunk_id(const char p[4])
{
int i;
for (i = 0; i < 4; i++)
if (p[i] < 0x20 || p[i] > 0x7f)
return sox_false;
return sox_true;
}
static int read_chunk_header(sox_format_t *ft, char tag[4], uint32_t *len)
{
int r;
r = lsx_readbuf(ft, tag, 4);
if (r < 4)
return SOX_EOF;
return lsx_readdw(ft, len);
}
/*
* Do anything required before you start reading samples.
* Read file header.
* Find out sampling rate,
* size and encoding of samples,
* mono/stereo/quad.
*/
static int startread(sox_format_t *ft)
{
priv_t *wav = ft->priv;
char magic[5] = { 0 };
uint32_t clen;
int err;
sox_bool isRF64 = sox_false;
uint64_t ds64_riff_size;
uint64_t ds64_data_size;
uint64_t ds64_sample_count;
/* wave file characteristics */
uint64_t qwRiffLength;
uint64_t qwDataLength = 0;
sox_bool have_fmt = sox_false;
ft->sox_errno = SOX_SUCCESS;
wav->ignoreSize = ft->signal.length == SOX_IGNORE_LENGTH;
ft->encoding.reverse_bytes = MACHINE_IS_BIGENDIAN;
if (read_chunk_header(ft, magic, &clen))
return SOX_EOF;
if (!memcmp(magic, "RIFX", 4)) {
lsx_debug("Found RIFX header");
ft->encoding.reverse_bytes = MACHINE_IS_LITTLEENDIAN;
} else if (!memcmp(magic, "RF64", 4)) {
lsx_debug("Found RF64 header");
isRF64 = sox_true;
} else if (memcmp(magic, "RIFF", 4)) {
lsx_fail_errno(ft, SOX_EHDR, "WAVE: RIFF header not found");
return SOX_EOF;
}
qwRiffLength = clen;
if (lsx_readbuf(ft, magic, 4) < 4 || memcmp(magic, "WAVE", 4)) {
lsx_fail_errno(ft, SOX_EHDR, "WAVE header not found");
return SOX_EOF;
}
while (!read_chunk_header(ft, magic, &clen)) {
uint32_t len = clen;
off_t cstart = lsx_tell(ft);
off_t pos;
if (!valid_chunk_id(magic)) {
lsx_fail_errno(ft, SOX_EHDR, "invalid chunk ID found");
return SOX_EOF;
}
lsx_debug("Found chunk '%s', size %u", magic, clen);
if (!memcmp(magic, "ds64", 4)) {
if (!isRF64)
lsx_warn("ds64 chunk in non-RF64 file");
if (clen < 28) {
lsx_fail_errno(ft, SOX_EHDR, "ds64 chunk too small");
return SOX_EOF;
}
if (clen == 32) {
lsx_warn("ds64 chunk size invalid, attempting workaround");
clen = 28;
}
err = lsx_read_fields(ft, &len, "qqq",
&ds64_riff_size,
&ds64_data_size,
&ds64_sample_count);
if (err)
return SOX_EOF;
goto next;
}
if (!memcmp(magic, "fmt ", 4)) {
err = wav_read_fmt(ft, clen);
if (err)
return err;
have_fmt = sox_true;
goto next;
}
if (!memcmp(magic, "fact", 4)) {
uint32_t val;
err = lsx_read_fields(ft, &len, "i", &val);
if (err)
return SOX_EOF;
wav->numSamples = val;
goto next;
}
if (!memcmp(magic, "data", 4)) {
if (isRF64 && clen == UINT32_MAX)
clen = ds64_data_size;
qwDataLength = clen;
wav->dataStart = lsx_tell(ft);
if (qwDataLength == UINT32_MAX || qwDataLength == MS_UNSPEC)
break;
if (!ft->seekable)
break;
goto next;
}
next:
pos = lsx_tell(ft);
clen += clen & 1;
if (pos > cstart + clen) {
lsx_fail_errno(ft, SOX_EHDR, "malformed chunk %s", magic);
return SOX_EOF;
}
err = lsx_seeki(ft, cstart + clen - pos, SEEK_CUR);
if (err)
return SOX_EOF;
}
if (isRF64) {
if (wav->numSamples == UINT32_MAX)
wav->numSamples = ds64_sample_count;
if (qwRiffLength == UINT32_MAX)
qwRiffLength = ds64_riff_size;
}
if (!have_fmt) {
lsx_fail_errno(ft, SOX_EOF, "fmt chunk not found");
return SOX_EOF;
}
if (!wav->dataStart) {
lsx_fail_errno(ft, SOX_EOF, "data chunk not found");
return SOX_EOF;
}
if (ft->seekable)
lsx_seeki(ft, wav->dataStart, SEEK_SET);
/* some files wrongly report total samples across all channels */
if (wav->numSamples * wav->blockAlign == qwDataLength * ft->signal.channels)
wav->numSamples /= ft->signal.channels;
if ((qwDataLength == UINT32_MAX && !wav->numSamples) ||
qwDataLength == MS_UNSPEC) {
lsx_warn("WAV data length is magic value or UINT32_MAX, ignoring");
wav->ignoreSize = 1;
}
switch (wav->formatTag) {
case WAVE_FORMAT_ADPCM:
wav->numSamples =
lsx_ms_adpcm_samples_in(qwDataLength, ft->signal.channels,
wav->blockAlign, wav->samplesPerBlock);
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
break;
case WAVE_FORMAT_IMA_ADPCM:
/* Compute easiest part of number of samples. For every block, there
are samplesPerBlock samples to read. */
wav->numSamples =
lsx_ima_samples_in(qwDataLength, ft->signal.channels,
wav->blockAlign, wav->samplesPerBlock);
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
lsx_ima_init_table();
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
wav->numSamples = qwDataLength / wav->blockAlign * wav->samplesPerBlock;
wavgsminit(ft);
break;
#endif
}
if (!wav->numSamples)
wav->numSamples = div_bits(qwDataLength, ft->encoding.bits_per_sample)
/ ft->signal.channels;
if (wav->ignoreSize)
ft->signal.length = SOX_UNSPEC;
else
ft->signal.length = wav->numSamples * ft->signal.channels;
return lsx_rawstartread(ft);
}
/*
* Read up to len samples from file.
* Convert to signed longs.
* Place in buf[].
* Return number of samples read.
*/
static size_t read_samples(sox_format_t *ft, sox_sample_t *buf, size_t len)
{
priv_t *wav = ft->priv;
size_t done;
ft->sox_errno = SOX_SUCCESS;
if (!wav->ignoreSize)
len = min(len, wav->numSamples * ft->signal.channels);
/* If file is in ADPCM encoding then read in multiple blocks else */
/* read as much as possible and return quickly. */
switch (ft->encoding.encoding) {
case SOX_ENCODING_IMA_ADPCM:
case SOX_ENCODING_MS_ADPCM:
done = 0;
while (done < len) { /* Still want data? */
short *p, *top;
size_t ct;
/* See if need to read more from disk */
if (wav->blockSamplesRemaining == 0) {
if (wav->formatTag == WAVE_FORMAT_IMA_ADPCM)
wav->blockSamplesRemaining = ImaAdpcmReadBlock(ft);
else
wav->blockSamplesRemaining = AdpcmReadBlock(ft);
if (wav->blockSamplesRemaining == 0) {
/* Don't try to read any more samples */
wav->numSamples = 0;
return done;
}
wav->samplePtr = wav->samples;
}
/* Copy interleaved data into buf, converting to sox_sample_t */
ct = len - done;
if (ct > wav->blockSamplesRemaining * ft->signal.channels)
ct = wav->blockSamplesRemaining * ft->signal.channels;
done += ct;
wav->blockSamplesRemaining -= ct / ft->signal.channels;
p = wav->samplePtr;
top = p + ct;
/* Output is already signed */
while (p < top)
*buf++ = SOX_SIGNED_16BIT_TO_SAMPLE(*p++,);
wav->samplePtr = p;
}
/* "done" for ADPCM equals total data processed and not
* total samples procesed. The only way to take care of that
* is to return here and not fall thru.
*/
wav->numSamples -= done / ft->signal.channels;
return done;
#ifdef HAVE_LIBGSM
case SOX_ENCODING_GSM:
done = wavgsmread(ft, buf, len);
break;
#endif
default: /* assume PCM or float encoding */
done = lsx_rawread(ft, buf, len);
break;
}
if (done == 0 && wav->numSamples && !wav->ignoreSize)
lsx_warn("Premature EOF on .wav input file");
/* Only return buffers that contain a totally playable
* amount of audio.
*/
done -= done % ft->signal.channels;
if (done / ft->signal.channels > wav->numSamples)
wav->numSamples = 0;
else
wav->numSamples -= done / ft->signal.channels;
return done;
}
/*
* Do anything required when you stop reading samples.
* Don't close input file!
*/
static int stopread(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
ft->sox_errno = SOX_SUCCESS;
free(wav->packet);
free(wav->samples);
free(wav->lsx_ms_adpcm_i_coefs);
free(wav->ms_adpcm_data);
switch (ft->encoding.encoding)
{
#ifdef HAVE_LIBGSM
case SOX_ENCODING_GSM:
wavgsmdestroy(ft);
break;
#endif
case SOX_ENCODING_IMA_ADPCM:
case SOX_ENCODING_MS_ADPCM:
break;
default:
break;
}
return SOX_SUCCESS;
}
static int startwrite(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
int rc;
ft->sox_errno = SOX_SUCCESS;
if (ft->encoding.encoding != SOX_ENCODING_MS_ADPCM &&
ft->encoding.encoding != SOX_ENCODING_IMA_ADPCM &&
ft->encoding.encoding != SOX_ENCODING_GSM)
{
rc = lsx_rawstartwrite(ft);
if (rc)
return rc;
}
wav->numSamples = 0;
wav->dataLength = 0;
if (!ft->signal.length && !ft->seekable)
lsx_warn("Length in output .wav header will be wrong since can't seek to fix it");
rc = wavwritehdr(ft, 0); /* also calculates various wav->* info */
if (rc != 0)
return rc;
wav->packet = NULL;
wav->samples = NULL;
wav->lsx_ms_adpcm_i_coefs = NULL;
switch (wav->formatTag)
{
size_t ch, sbsize;
case WAVE_FORMAT_IMA_ADPCM:
lsx_ima_init_table();
/* intentional case fallthru! */
case WAVE_FORMAT_ADPCM:
/* #channels already range-checked for overflow in wavwritehdr() */
for (ch=0; ch<ft->signal.channels; ch++)
wav->state[ch] = 0;
sbsize = ft->signal.channels * wav->samplesPerBlock;
wav->packet = lsx_malloc((size_t)wav->blockAlign);
wav->samples = lsx_malloc(sbsize*sizeof(short));
wav->sampleTop = wav->samples + sbsize;
wav->samplePtr = wav->samples;
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
return wavgsminit(ft);
#endif
default:
break;
}
return SOX_SUCCESS;
}
/* wavwritehdr: write .wav headers as follows:
bytes variable description
0 - 3 'RIFF'/'RIFX' Little/Big-endian
4 - 7 wRiffLength length of file minus the 8 byte riff header
8 - 11 'WAVE'
12 - 15 'fmt '
16 - 19 wFmtSize length of format chunk minus 8 byte header
20 - 21 wFormatTag identifies PCM, ULAW etc
22 - 23 wChannels
24 - 27 dwSamplesPerSecond samples per second per channel
28 - 31 dwAvgBytesPerSec non-trivial for compressed formats
32 - 33 wBlockAlign basic block size
34 - 35 wBitsPerSample non-trivial for compressed formats
PCM formats then go straight to the data chunk:
36 - 39 'data'
40 - 43 dwDataLength length of data chunk minus 8 byte header
44 - (dwDataLength + 43) the data
(+ a padding byte if dwDataLength is odd)
non-PCM formats must write an extended format chunk and a fact chunk:
ULAW, ALAW formats:
36 - 37 wExtSize = 0 the length of the format extension
38 - 41 'fact'
42 - 45 dwFactSize = 4 length of the fact chunk minus 8 byte header
46 - 49 dwSamplesWritten actual number of samples written out
50 - 53 'data'
54 - 57 dwDataLength length of data chunk minus 8 byte header
58 - (dwDataLength + 57) the data
(+ a padding byte if dwDataLength is odd)
GSM6.10 format:
36 - 37 wExtSize = 2 the length in bytes of the format-dependent extension
38 - 39 320 number of samples per block
40 - 43 'fact'
44 - 47 dwFactSize = 4 length of the fact chunk minus 8 byte header
48 - 51 dwSamplesWritten actual number of samples written out
52 - 55 'data'
56 - 59 dwDataLength length of data chunk minus 8 byte header
60 - (dwDataLength + 59) the data (including a padding byte, if necessary,
so dwDataLength is always even)
note that header contains (up to) 3 separate ways of describing the
length of the file, all derived here from the number of (input)
samples wav->numSamples in a way that is non-trivial for the blocked
and padded compressed formats:
wRiffLength - (riff header) the length of the file, minus 8
dwSamplesWritten - (fact header) the number of samples written (after padding
to a complete block eg for GSM)
dwDataLength - (data chunk header) the number of (valid) data bytes written
*/
static int wavwritehdr(sox_format_t * ft, int second_header)
{
priv_t * wav = (priv_t *) ft->priv;
/* variables written to wav file header */
/* RIFF header */
uint64_t wRiffLength ; /* length of file after 8 byte riff header */
/* fmt chunk */
uint16_t wFmtSize = 16; /* size field of the fmt chunk */
uint16_t wFormatTag = 0; /* data format */
uint16_t wChannels; /* number of channels */
uint32_t dwSamplesPerSecond; /* samples per second per channel*/
uint32_t dwAvgBytesPerSec=0; /* estimate of bytes per second needed */
uint32_t wBlockAlign=0; /* byte alignment of a basic sample block */
uint16_t wBitsPerSample=0; /* bits per sample */
/* fmt chunk extension (not PCM) */
uint16_t wExtSize=0; /* extra bytes in the format extension */
uint16_t wSamplesPerBlock; /* samples per channel per block */
/* wSamplesPerBlock and other things may go into format extension */
/* fact chunk (not PCM) */
uint32_t dwFactSize=4; /* length of the fact chunk */
uint64_t dwSamplesWritten=0; /* windows doesnt seem to use this*/
/* data chunk */
uint64_t dwDataLength; /* length of sound data in bytes */
/* end of variables written to header */
/* internal variables, intermediate values etc */
int bytespersample; /* (uncompressed) bytes per sample (per channel) */
uint64_t blocksWritten = 0;
sox_bool isExtensible = sox_false; /* WAVE_FORMAT_EXTENSIBLE? */
if (ft->signal.channels > UINT16_MAX) {
lsx_fail_errno(ft, SOX_EOF, "Too many channels (%u)",
ft->signal.channels);
return SOX_EOF;
}
dwSamplesPerSecond = ft->signal.rate;
wChannels = ft->signal.channels;
wBitsPerSample = ft->encoding.bits_per_sample;
wSamplesPerBlock = 1; /* common default for PCM data */
switch (ft->encoding.encoding)
{
case SOX_ENCODING_UNSIGNED:
case SOX_ENCODING_SIGN2:
wFormatTag = WAVE_FORMAT_PCM;
bytespersample = (wBitsPerSample + 7)/8;
wBlockAlign = wChannels * bytespersample;
break;
case SOX_ENCODING_FLOAT:
wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
bytespersample = (wBitsPerSample + 7)/8;
wBlockAlign = wChannels * bytespersample;
break;
case SOX_ENCODING_ALAW:
wFormatTag = WAVE_FORMAT_ALAW;
wBlockAlign = wChannels;
break;
case SOX_ENCODING_ULAW:
wFormatTag = WAVE_FORMAT_MULAW;
wBlockAlign = wChannels;
break;
case SOX_ENCODING_IMA_ADPCM:
if (wChannels>16)
{
lsx_fail_errno(ft,SOX_EOF,"Channels(%d) must be <= 16",wChannels);
return SOX_EOF;
}
wFormatTag = WAVE_FORMAT_IMA_ADPCM;
wBlockAlign = wChannels * 256; /* reasonable default */
wBitsPerSample = 4;
wExtSize = 2;
wSamplesPerBlock = lsx_ima_samples_in((size_t) 0, (size_t) wChannels, (size_t) wBlockAlign, (size_t) 0);
break;
case SOX_ENCODING_MS_ADPCM:
if (wChannels>16)
{
lsx_fail_errno(ft,SOX_EOF,"Channels(%d) must be <= 16",wChannels);
return SOX_EOF;
}
wFormatTag = WAVE_FORMAT_ADPCM;
wBlockAlign = ft->signal.rate / 11008;
wBlockAlign = max(wBlockAlign, 1) * wChannels * 256;
wBitsPerSample = 4;
wExtSize = 4+4*7; /* Ext fmt data length */
wSamplesPerBlock = lsx_ms_adpcm_samples_in((size_t) 0, (size_t) wChannels, (size_t) wBlockAlign, (size_t) 0);
break;
#ifdef HAVE_LIBGSM
case SOX_ENCODING_GSM:
if (wChannels!=1)
{
lsx_report("Overriding GSM audio from %d channel to 1",wChannels);
if (!second_header)
ft->signal.length /= max(1, ft->signal.channels);
wChannels = ft->signal.channels = 1;
}
wFormatTag = WAVE_FORMAT_GSM610;
/* dwAvgBytesPerSec = 1625*(dwSamplesPerSecond/8000.)+0.5; */
wBlockAlign=65;
wBitsPerSample=0; /* not representable as int */
wExtSize=2; /* length of format extension */
wSamplesPerBlock = 320;
break;
#endif
default:
break;
}
if (wBlockAlign > UINT16_MAX) {
lsx_fail_errno(ft, SOX_EOF, "Too many channels (%u)",
ft->signal.channels);
return SOX_EOF;
}
wav->formatTag = wFormatTag;
wav->blockAlign = wBlockAlign;
wav->samplesPerBlock = wSamplesPerBlock;
/* When creating header, use length hint given by input file. If no
* hint then write default value. Also, use default value even
* on header update if more then 32-bit length needs to be written.
*/
dwSamplesWritten =
second_header ? wav->numSamples : ft->signal.length / wChannels;
blocksWritten =
(dwSamplesWritten + wSamplesPerBlock - 1) / wSamplesPerBlock;
dwDataLength = blocksWritten * wBlockAlign;
if (wFormatTag == WAVE_FORMAT_GSM610)
dwDataLength = (dwDataLength+1) & ~1u; /* round up to even */
if (wFormatTag == WAVE_FORMAT_PCM && (wBitsPerSample > 16 || wChannels > 2)
&& strcmp(ft->filetype, "wavpcm")) {
isExtensible = sox_true;
wFmtSize += 2 + 22;
}
else if (wFormatTag != WAVE_FORMAT_PCM)
wFmtSize += 2+wExtSize; /* plus ExtData */
wRiffLength = 4 + (8+wFmtSize) + (8+dwDataLength+dwDataLength%2);
if (isExtensible || wFormatTag != WAVE_FORMAT_PCM) /* PCM omits the "fact" chunk */
wRiffLength += (8+dwFactSize);
if (dwSamplesWritten > UINT32_MAX)
dwSamplesWritten = UINT32_MAX;
if (dwDataLength > UINT32_MAX)
dwDataLength = UINT32_MAX;
if (!second_header && !ft->signal.length)
dwDataLength = UINT32_MAX;
if (wRiffLength > UINT32_MAX)
wRiffLength = UINT32_MAX;
/* dwAvgBytesPerSec <-- this is BEFORE compression, isn't it? guess not. */
dwAvgBytesPerSec = (double)wBlockAlign*ft->signal.rate / (double)wSamplesPerBlock + 0.5;
/* figured out header info, so write it */
/* If user specified opposite swap than we think, assume they are
* asking to write a RIFX file.
*/
if (ft->encoding.reverse_bytes == MACHINE_IS_LITTLEENDIAN)
{
if (!second_header)
lsx_report("Requested to swap bytes so writing RIFX header");
lsx_writes(ft, "RIFX");
}
else
lsx_writes(ft, "RIFF");
lsx_writedw(ft, wRiffLength);
lsx_writes(ft, "WAVE");
lsx_writes(ft, "fmt ");
lsx_writedw(ft, wFmtSize);
lsx_writew(ft, isExtensible ? WAVE_FORMAT_EXTENSIBLE : wFormatTag);
lsx_writew(ft, wChannels);
lsx_writedw(ft, dwSamplesPerSecond);
lsx_writedw(ft, dwAvgBytesPerSec);
lsx_writew(ft, wBlockAlign);
lsx_writew(ft, wBitsPerSample); /* end info common to all fmts */
if (isExtensible) {
uint32_t dwChannelMask=0; /* unassigned speaker mapping by default */
static unsigned char const guids[][14] = {
"\x00\x00\x00\x00\x10\x00\x80\x00\x00\xAA\x00\x38\x9B\x71", /* wav */
"\x00\x00\x21\x07\xd3\x11\x86\x44\xc8\xc1\xca\x00\x00\x00"}; /* amb */
/* if not amb, assume most likely channel masks from number of channels; not
* ideal solution, but will make files playable in many/most situations
*/
if (strcmp(ft->filetype, "amb")) {
if (wChannels == 1) dwChannelMask = 0x4; /* 1 channel (mono) = FC */
else if (wChannels == 2) dwChannelMask = 0x3; /* 2 channels (stereo) = FL, FR */
else if (wChannels == 4) dwChannelMask = 0x33; /* 4 channels (quad) = FL, FR, BL, BR */
else if (wChannels == 6) dwChannelMask = 0x3F; /* 6 channels (5.1) = FL, FR, FC, LF, BL, BR */
else if (wChannels == 8) dwChannelMask = 0x63F; /* 8 channels (7.1) = FL, FR, FC, LF, BL, BR, SL, SR */
}
lsx_writew(ft, 22);
lsx_writew(ft, wBitsPerSample); /* No padding in container */
lsx_writedw(ft, dwChannelMask); /* Speaker mapping is something reasonable */
lsx_writew(ft, wFormatTag);
lsx_writebuf(ft, guids[!strcmp(ft->filetype, "amb")], (size_t)14);
}
else
/* if not PCM, we need to write out wExtSize even if wExtSize=0 */
if (wFormatTag != WAVE_FORMAT_PCM)
lsx_writew(ft,wExtSize);
switch (wFormatTag)
{
int i;
case WAVE_FORMAT_IMA_ADPCM:
lsx_writew(ft, wSamplesPerBlock);
break;
case WAVE_FORMAT_ADPCM:
lsx_writew(ft, wSamplesPerBlock);
lsx_writew(ft, 7); /* nCoefs */
for (i=0; i<7; i++) {
lsx_writew(ft, (uint16_t)(lsx_ms_adpcm_i_coef[i][0]));
lsx_writew(ft, (uint16_t)(lsx_ms_adpcm_i_coef[i][1]));
}
break;
case WAVE_FORMAT_GSM610:
lsx_writew(ft, wSamplesPerBlock);
break;
default:
break;
}
/* if not PCM, write the 'fact' chunk */
if (isExtensible || wFormatTag != WAVE_FORMAT_PCM){
lsx_writes(ft, "fact");
lsx_writedw(ft,dwFactSize);
lsx_writedw(ft,dwSamplesWritten);
}
lsx_writes(ft, "data");
lsx_writedw(ft, dwDataLength); /* data chunk size */
if (!second_header) {
lsx_debug("Writing Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wFormatTag), wChannels,
wChannels == 1 ? "" : "s", dwSamplesPerSecond);
lsx_debug(" %d byte/sec, %d block align, %d bits/samp",
dwAvgBytesPerSec, wBlockAlign, wBitsPerSample);
} else {
if (wRiffLength == UINT32_MAX || dwDataLength == UINT32_MAX ||
dwSamplesWritten == UINT32_MAX)
lsx_warn("File too large, writing truncated values in header");
lsx_debug("Finished writing Wave file, %"PRIu64" data bytes %"PRIu64" samples",
dwDataLength, wav->numSamples);
#ifdef HAVE_LIBGSM
if (wFormatTag == WAVE_FORMAT_GSM610){
lsx_debug("GSM6.10 format: %"PRIu64" blocks %"PRIu64" padded samples %"PRIu64" padded data bytes",
blocksWritten, dwSamplesWritten, dwDataLength);
if (wav->gsmbytecount != dwDataLength)
lsx_warn("help ! internal inconsistency - data_written %"PRIu64" gsmbytecount %zu",
dwDataLength, wav->gsmbytecount);
}
#endif
}
return SOX_SUCCESS;
}
static size_t write_samples(sox_format_t * ft, const sox_sample_t *buf, size_t len)
{
priv_t * wav = (priv_t *) ft->priv;
ptrdiff_t total_len = len;
ft->sox_errno = SOX_SUCCESS;
switch (wav->formatTag)
{
case WAVE_FORMAT_IMA_ADPCM:
case WAVE_FORMAT_ADPCM:
while (len>0) {
short *p = wav->samplePtr;
short *top = wav->sampleTop;
if (top>p+len) top = p+len;
len -= top-p; /* update residual len */
while (p < top)
*p++ = (*buf++) >> 16;
wav->samplePtr = p;
if (p == wav->sampleTop)
xxxAdpcmWriteBlock(ft);
}
return total_len - len;
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
len = wavgsmwrite(ft, buf, len);
wav->numSamples += (len/ft->signal.channels);
return len;
break;
#endif
default:
len = lsx_rawwrite(ft, buf, len);
wav->numSamples += (len/ft->signal.channels);
return len;
}
}
static int stopwrite(sox_format_t * ft)
{
priv_t * wav = (priv_t *) ft->priv;
ft->sox_errno = SOX_SUCCESS;
/* Call this to flush out any remaining data. */
switch (wav->formatTag)
{
case WAVE_FORMAT_IMA_ADPCM:
case WAVE_FORMAT_ADPCM:
xxxAdpcmWriteBlock(ft);
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
wavgsmstopwrite(ft);
break;
#endif
}
/* Add a pad byte if the number of data bytes is odd.
See wavwritehdr() above for the calculation. */
if (wav->formatTag != WAVE_FORMAT_GSM610)
lsx_padbytes(ft, (size_t)((wav->numSamples + wav->samplesPerBlock - 1)/wav->samplesPerBlock*wav->blockAlign) % 2);
free(wav->packet);
free(wav->samples);
free(wav->lsx_ms_adpcm_i_coefs);
/* All samples are already written out. */
/* If file header needs fixing up, for example it needs the */
/* the number of samples in a field, seek back and write them here. */
if (ft->signal.length && wav->numSamples <= 0xffffffff &&
wav->numSamples == ft->signal.length)
return SOX_SUCCESS;
if (!ft->seekable)
return SOX_EOF;
if (lsx_seeki(ft, (off_t)0, SEEK_SET) != 0)
{
lsx_fail_errno(ft,SOX_EOF,"Can't rewind output file to rewrite .wav header.");
return SOX_EOF;
}
return (wavwritehdr(ft, 1));
}
/*
* Return a string corresponding to the wave format type.
*/
static const char *wav_format_str(unsigned tag)
{
const struct wave_format *f = wav_find_format(tag);
return f ? f->name : "unknown";
}
static int seek(sox_format_t * ft, uint64_t offset)
{
priv_t * wav = (priv_t *) ft->priv;
if (ft->encoding.bits_per_sample & 7)
lsx_fail_errno(ft, SOX_ENOTSUP, "seeking not supported with this encoding");
else if (wav->formatTag == WAVE_FORMAT_GSM610) {
int alignment;
size_t gsmoff;
/* rounding bytes to blockAlign so that we
* don't have to decode partial block. */
gsmoff = offset * wav->blockAlign / wav->samplesPerBlock +
wav->blockAlign * ft->signal.channels / 2;
gsmoff -= gsmoff % (wav->blockAlign * ft->signal.channels);
ft->sox_errno = lsx_seeki(ft, (off_t)(gsmoff + wav->dataStart), SEEK_SET);
if (ft->sox_errno == SOX_SUCCESS) {
/* offset is in samples */
uint64_t new_offset = offset;
alignment = offset % wav->samplesPerBlock;
if (alignment != 0)
new_offset += (wav->samplesPerBlock - alignment);
wav->numSamples = ft->signal.length - (new_offset / ft->signal.channels);
}
} else {
double wide_sample = offset - (offset % ft->signal.channels);
double to_d = wide_sample * ft->encoding.bits_per_sample / 8;
off_t to = to_d;
ft->sox_errno = (to != to_d)? SOX_EOF : lsx_seeki(ft, (off_t)wav->dataStart + (off_t)to, SEEK_SET);
if (ft->sox_errno == SOX_SUCCESS)
wav->numSamples -= (size_t)wide_sample / ft->signal.channels;
}
return ft->sox_errno;
}
LSX_FORMAT_HANDLER(wav)
{
static char const * const names[] = {"wav", "wavpcm", "amb", NULL};
static unsigned const write_encodings[] = {
SOX_ENCODING_SIGN2, 16, 24, 32, 0,
SOX_ENCODING_UNSIGNED, 8, 0,
SOX_ENCODING_ULAW, 8, 0,
SOX_ENCODING_ALAW, 8, 0,
#ifdef HAVE_LIBGSM
SOX_ENCODING_GSM, 0,
#endif
SOX_ENCODING_MS_ADPCM, 4, 0,
SOX_ENCODING_IMA_ADPCM, 4, 0,
SOX_ENCODING_FLOAT, 32, 64, 0,
0};
static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE,
"Microsoft audio format", names, SOX_FILE_LIT_END,
startread, read_samples, stopread,
startwrite, write_samples, stopwrite,
seek, write_encodings, NULL, sizeof(priv_t)
};
return &handler;
}