ref: 6fa0f577dab703fc3c682eb8ab604aa6912a4577
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 EXTERNAL_GSM
#ifdef HAVE_GSM_GSM_H
#include <gsm/gsm.h>
#else
#include <gsm.h>
#endif
#else
#include "../libgsm/gsm.h"
#endif
#define WAVE_FORMAT_UNKNOWN (0x0000U)
#define WAVE_FORMAT_PCM (0x0001U)
#define WAVE_FORMAT_ADPCM (0x0002U)
#define WAVE_FORMAT_IEEE_FLOAT (0x0003U)
#define WAVE_FORMAT_ALAW (0x0006U)
#define WAVE_FORMAT_MULAW (0x0007U)
#define WAVE_FORMAT_OKI_ADPCM (0x0010U)
#define WAVE_FORMAT_IMA_ADPCM (0x0011U)
#define WAVE_FORMAT_DIGISTD (0x0015U)
#define WAVE_FORMAT_DIGIFIX (0x0016U)
#define WAVE_FORMAT_DOLBY_AC2 (0x0030U)
#define WAVE_FORMAT_GSM610 (0x0031U)
#define WAVE_FORMAT_ROCKWELL_ADPCM (0x003bU)
#define WAVE_FORMAT_ROCKWELL_DIGITALK (0x003cU)
#define WAVE_FORMAT_G721_ADPCM (0x0040U)
#define WAVE_FORMAT_G728_CELP (0x0041U)
#define WAVE_FORMAT_MPEG (0x0050U)
#define WAVE_FORMAT_MPEGLAYER3 (0x0055U)
#define WAVE_FORMAT_G726_ADPCM (0x0064U)
#define WAVE_FORMAT_G722_ADPCM (0x0065U)
#define WAVE_FORMAT_EXTENSIBLE (0xfffeU)
/* To allow padding to samplesPerBlock. Works, but currently never true. */
static size_t pad_nsamps = sox_false;
/* Private data for .wav file */
typedef struct {
size_t numSamples; /* samples/channel reading: starts at total count and decremented */
/* writing: starts at 0 and counts samples written */
size_t dataLength; /* needed for ADPCM writing */
unsigned short formatTag; /* What type of encoding file is using */
unsigned short samplesPerBlock;
unsigned short blockAlign;
size_t dataStart; /* need to for seeking */
char * comment;
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 */
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 */
/* following used by GSM 6.10 wav */
gsm gsmhandle;
gsm_signal *gsmsample;
int gsmindex;
size_t gsmbytecount; /* counts bytes written to data block */
} priv_t;
static char *wav_format_str(unsigned wFormatTag);
static int wavwritehdr(sox_format_t *, int);
/****************************************************************************/
/* IMA ADPCM Support Functions Section */
/****************************************************************************/
/*
*
* 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)
{
sox_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 */
/****************************************************************************/
/*
*
* 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)
{
sox_warn("Premature EOF on .wav input file");
return 0;
}
}
errmsg = lsx_ms_adpcm_block_expand_i(ft->signal.channels, wav->nCoefs, wav->lsx_ms_adpcm_i_coefs, wav->packet, wav->samples, samplesThisBlock);
if (errmsg)
sox_warn(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);
}
/****************************************************************************/
/* WAV GSM6.10 support functions */
/****************************************************************************/
/* 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) {
sox_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) {
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);
}
/****************************************************************************/
/* General Sox WAV file code */
/****************************************************************************/
static int findChunk(sox_format_t * ft, const char *Label, uint32_t *len)
{
char magic[5];
for (;;)
{
if (lsx_reads(ft, magic, (size_t)4) == SOX_EOF)
{
lsx_fail_errno(ft, SOX_EHDR, "WAVE file has missing %s chunk",
Label);
return SOX_EOF;
}
sox_debug("WAV Chunk %s", magic);
if (lsx_readdw(ft, len) == SOX_EOF)
{
lsx_fail_errno(ft, SOX_EHDR, "WAVE file %s chunk is too short",
magic);
return SOX_EOF;
}
if (strncmp(Label, magic, (size_t)4) == 0)
break; /* Found the given chunk */
/* skip to next chunk */
if (*len == 0 || lsx_seeki(ft, (ptrdiff_t)(*len), SEEK_CUR) != SOX_SUCCESS)
{
lsx_fail_errno(ft,SOX_EHDR,
"WAV chunk appears to have invalid size %d.", *len);
return SOX_EOF;
}
}
return SOX_SUCCESS;
}
static int wavfail(sox_format_t * ft, const char *format)
{
lsx_fail_errno(ft, SOX_EHDR, "Unhandled WAV file encoding (%s).\nTry overriding the encoding: e.g. for an MP3 WAV, `-t mp3'", format);
return SOX_EOF;
}
/*
* 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 = (priv_t *) ft->priv;
char magic[5];
uint32_t len;
/* wave file characteristics */
uint32_t dwRiffLength;
unsigned short wChannels; /* number of channels */
uint32_t dwSamplesPerSecond; /* samples per second per channel */
uint32_t dwAvgBytesPerSec;/* estimate of bytes per second needed */
uint16_t wBitsPerSample; /* bits per sample */
uint32_t wFmtSize;
uint16_t wExtSize = 0; /* extended field for non-PCM */
uint32_t dwDataLength; /* length of sound data in bytes */
size_t bytesPerBlock = 0;
int bytespersample; /* bytes per sample (per channel */
char text[256];
uint32_t dwLoopPos;
ft->sox_errno = SOX_SUCCESS;
wav->ignoreSize = 0;
if (lsx_reads(ft, magic, (size_t)4) == SOX_EOF || (strncmp("RIFF", magic, (size_t)4) != 0 &&
strncmp("RIFX", magic, (size_t)4) != 0))
{
lsx_fail_errno(ft,SOX_EHDR,"WAVE: RIFF header not found");
return SOX_EOF;
}
/* RIFX is a Big-endian RIFF */
if (strncmp("RIFX", magic, (size_t)4) == 0)
{
sox_debug("Found RIFX header");
ft->encoding.reverse_bytes = MACHINE_IS_LITTLEENDIAN;
}
else ft->encoding.reverse_bytes = MACHINE_IS_BIGENDIAN;
lsx_readdw(ft, &dwRiffLength);
if (lsx_reads(ft, magic, (size_t)4) == SOX_EOF || strncmp("WAVE", magic, (size_t)4))
{
lsx_fail_errno(ft,SOX_EHDR,"WAVE header not found");
return SOX_EOF;
}
/* Now look for the format chunk */
if (findChunk(ft, "fmt ", &len) == SOX_EOF)
{
lsx_fail_errno(ft,SOX_EHDR,"WAVE chunk fmt not found");
return SOX_EOF;
}
wFmtSize = len;
if (wFmtSize < 16)
{
lsx_fail_errno(ft,SOX_EHDR,"WAVE file fmt chunk is too short");
return SOX_EOF;
}
lsx_readw(ft, &(wav->formatTag));
lsx_readw(ft, &wChannels);
lsx_readdw(ft, &dwSamplesPerSecond);
lsx_readdw(ft, &dwAvgBytesPerSec); /* Average bytes/second */
lsx_readw(ft, &(wav->blockAlign)); /* Block align */
lsx_readw(ft, &wBitsPerSample); /* bits per sample per channel */
len -= 16;
if (wav->formatTag == WAVE_FORMAT_EXTENSIBLE)
{
uint16_t extensionSize;
uint16_t numberOfValidBits;
uint32_t speakerPositionMask;
uint16_t subFormatTag;
uint8_t dummyByte;
int i;
if (wFmtSize < 18)
{
lsx_fail_errno(ft,SOX_EHDR,"WAVE file fmt chunk is too short");
return SOX_EOF;
}
lsx_readw(ft, &extensionSize);
len -= 2;
if (extensionSize < 22)
{
lsx_fail_errno(ft,SOX_EHDR,"WAVE file fmt chunk is too short");
return SOX_EOF;
}
lsx_readw(ft, &numberOfValidBits);
lsx_readdw(ft, &speakerPositionMask);
lsx_readw(ft, &subFormatTag);
for (i = 0; i < 14; ++i) lsx_readb(ft, &dummyByte);
len -= 22;
if (numberOfValidBits != wBitsPerSample)
{
lsx_fail_errno(ft,SOX_EHDR,"WAVE file fmt with padded samples is not supported yet");
return SOX_EOF;
}
wav->formatTag = subFormatTag;
}
switch (wav->formatTag)
{
case WAVE_FORMAT_UNKNOWN:
lsx_fail_errno(ft,SOX_EHDR,"WAVE file is in unsupported Microsoft Official Unknown format.");
return SOX_EOF;
case WAVE_FORMAT_PCM:
/* Default (-1) depends on sample size. Set that later on. */
if (ft->encoding.encoding != SOX_ENCODING_UNKNOWN && ft->encoding.encoding != SOX_ENCODING_UNSIGNED &&
ft->encoding.encoding != SOX_ENCODING_SIGN2)
sox_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_IMA_ADPCM:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN || ft->encoding.encoding == SOX_ENCODING_IMA_ADPCM)
ft->encoding.encoding = SOX_ENCODING_IMA_ADPCM;
else
sox_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_ADPCM:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN || ft->encoding.encoding == SOX_ENCODING_MS_ADPCM)
ft->encoding.encoding = SOX_ENCODING_MS_ADPCM;
else
sox_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_IEEE_FLOAT:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN || ft->encoding.encoding == SOX_ENCODING_FLOAT)
ft->encoding.encoding = SOX_ENCODING_FLOAT;
else
sox_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_ALAW:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN || ft->encoding.encoding == SOX_ENCODING_ALAW)
ft->encoding.encoding = SOX_ENCODING_ALAW;
else
sox_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_MULAW:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN || ft->encoding.encoding == SOX_ENCODING_ULAW)
ft->encoding.encoding = SOX_ENCODING_ULAW;
else
sox_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_OKI_ADPCM:
return wavfail(ft, "OKI ADPCM");
case WAVE_FORMAT_DIGISTD:
return wavfail(ft, "Digistd");
case WAVE_FORMAT_DIGIFIX:
return wavfail(ft, "Digifix");
case WAVE_FORMAT_DOLBY_AC2:
return wavfail(ft, "Dolby AC2");
case WAVE_FORMAT_GSM610:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN || ft->encoding.encoding == SOX_ENCODING_GSM )
ft->encoding.encoding = SOX_ENCODING_GSM;
else
sox_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_ROCKWELL_ADPCM:
return wavfail(ft, "Rockwell ADPCM");
case WAVE_FORMAT_ROCKWELL_DIGITALK:
return wavfail(ft, "Rockwell DIGITALK");
case WAVE_FORMAT_G721_ADPCM:
return wavfail(ft, "G.721 ADPCM");
case WAVE_FORMAT_G728_CELP:
return wavfail(ft, "G.728 CELP");
case WAVE_FORMAT_MPEG:
return wavfail(ft, "MPEG");
case WAVE_FORMAT_MPEGLAYER3:
return wavfail(ft, "MP3");
case WAVE_FORMAT_G726_ADPCM:
return wavfail(ft, "G.726 ADPCM");
case WAVE_FORMAT_G722_ADPCM:
return wavfail(ft, "G.722 ADPCM");
default:
lsx_fail_errno(ft, SOX_EHDR, "Unknown WAV file encoding (type %x)", wav->formatTag);
return SOX_EOF;
}
/* User options take precedence */
if (ft->signal.channels == 0 || ft->signal.channels == wChannels)
ft->signal.channels = wChannels;
else
sox_report("User options overriding channels read in .wav header");
if (ft->signal.rate == 0 || ft->signal.rate == dwSamplesPerSecond)
ft->signal.rate = dwSamplesPerSecond;
else
sox_report("User options overriding rate read in .wav header");
wav->lsx_ms_adpcm_i_coefs = NULL;
wav->packet = NULL;
wav->samples = NULL;
/* non-PCM formats expect 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) {
if (len >= 2) {
lsx_readw(ft, &wExtSize);
len -= 2;
} else {
sox_warn("wave header missing FmtExt chunk");
}
}
if (wExtSize > len)
{
lsx_fail_errno(ft,SOX_EOF,"wave header error: wExtSize inconsistent with wFmtLen");
return SOX_EOF;
}
switch (wav->formatTag)
{
case WAVE_FORMAT_ADPCM:
if (wExtSize < 4)
{
lsx_fail_errno(ft,SOX_EOF,"format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 4);
return SOX_EOF;
}
if (wBitsPerSample != 4)
{
lsx_fail_errno(ft,SOX_EOF,"Can only handle 4-bit MS ADPCM in wav files");
return SOX_EOF;
}
lsx_readw(ft, &(wav->samplesPerBlock));
bytesPerBlock = lsx_ms_adpcm_bytes_per_block((size_t) ft->signal.channels, (size_t) wav->samplesPerBlock);
if (bytesPerBlock > wav->blockAlign)
{
lsx_fail_errno(ft,SOX_EOF,"format[%s]: samplesPerBlock(%d) incompatible with blockAlign(%d)",
wav_format_str(wav->formatTag), wav->samplesPerBlock, wav->blockAlign);
return SOX_EOF;
}
lsx_readw(ft, &(wav->nCoefs));
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;
}
wav->packet = lsx_malloc((size_t)wav->blockAlign);
len -= 4;
if (wExtSize < 4 + 4*wav->nCoefs)
{
lsx_fail_errno(ft,SOX_EOF,"wave header error: wExtSize(%d) too small for nCoefs(%d)", wExtSize, wav->nCoefs);
return SOX_EOF;
}
wav->samples = lsx_malloc(wChannels*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));
{
int i, errct=0;
for (i=0; len>=2 && i < 2*wav->nCoefs; i++) {
lsx_readsw(ft, &(wav->lsx_ms_adpcm_i_coefs[i]));
len -= 2;
if (i<14) errct += (wav->lsx_ms_adpcm_i_coefs[i] != lsx_ms_adpcm_i_coef[i/2][i%2]);
/* sox_debug("lsx_ms_adpcm_i_coefs[%2d] %4d",i,wav->lsx_ms_adpcm_i_coefs[i]); */
}
if (errct) sox_warn("base lsx_ms_adpcm_i_coefs differ in %d/14 positions",errct);
}
bytespersample = 2; /* AFTER de-compression */
break;
case WAVE_FORMAT_IMA_ADPCM:
if (wExtSize < 2)
{
lsx_fail_errno(ft,SOX_EOF,"format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 2);
return SOX_EOF;
}
if (wBitsPerSample != 4)
{
lsx_fail_errno(ft,SOX_EOF,"Can only handle 4-bit IMA ADPCM in wav files");
return SOX_EOF;
}
lsx_readw(ft, &(wav->samplesPerBlock));
bytesPerBlock = lsx_ima_bytes_per_block((size_t) ft->signal.channels, (size_t) wav->samplesPerBlock);
if (bytesPerBlock > wav->blockAlign || wav->samplesPerBlock%8 != 1)
{
lsx_fail_errno(ft,SOX_EOF,"format[%s]: samplesPerBlock(%d) incompatible with blockAlign(%d)",
wav_format_str(wav->formatTag), wav->samplesPerBlock, wav->blockAlign);
return SOX_EOF;
}
wav->packet = lsx_malloc((size_t)wav->blockAlign);
len -= 2;
wav->samples = lsx_malloc(wChannels*wav->samplesPerBlock*sizeof(short));
bytespersample = 2; /* AFTER de-compression */
break;
/* GSM formats have extended fmt chunk. Check for those cases. */
case WAVE_FORMAT_GSM610:
if (wExtSize < 2)
{
lsx_fail_errno(ft,SOX_EOF,"format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 2);
return SOX_EOF;
}
lsx_readw(ft, &wav->samplesPerBlock);
bytesPerBlock = 65;
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;
}
bytespersample = 2; /* AFTER de-compression */
len -= 2;
break;
default:
bytespersample = (wBitsPerSample + 7)/8;
}
/* User options take precedence */
if (!ft->encoding.bits_per_sample || ft->encoding.bits_per_sample == wBitsPerSample)
ft->encoding.bits_per_sample = wBitsPerSample;
else
sox_warn("User options overriding size read in .wav header");
/* Now we have enough information to set default encodings. */
switch (bytespersample)
{
case 1:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN)
ft->encoding.encoding = SOX_ENCODING_UNSIGNED;
break;
case 2: case 3: case 4:
if (ft->encoding.encoding == SOX_ENCODING_UNKNOWN)
ft->encoding.encoding = SOX_ENCODING_SIGN2;
break;
default:
lsx_fail_errno(ft,SOX_EFMT,"Sorry, don't understand .wav size");
return SOX_EOF;
}
/* Skip anything left over from fmt chunk */
lsx_seeki(ft, (ptrdiff_t)len, SEEK_CUR);
/* for non-PCM formats, there's a 'fact' chunk before
* the upcoming 'data' chunk */
/* Now look for the wave data chunk */
if (findChunk(ft, "data", &len) == SOX_EOF)
{
lsx_fail_errno(ft, SOX_EOF, "Could not find data chunk.");
return SOX_EOF;
}
dwDataLength = len;
/* Data starts here */
wav->dataStart = lsx_tell(ft);
switch (wav->formatTag)
{
case WAVE_FORMAT_ADPCM:
wav->numSamples =
lsx_ms_adpcm_samples_in((size_t)dwDataLength, (size_t)ft->signal.channels,
(size_t)wav->blockAlign, (size_t)wav->samplesPerBlock);
sox_debug_more("datalen %d, numSamples %lu",dwDataLength, (unsigned long)wav->numSamples);
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
ft->signal.length = wav->numSamples*ft->signal.channels;
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((size_t)dwDataLength, (size_t)ft->signal.channels,
(size_t)wav->blockAlign, (size_t)wav->samplesPerBlock);
sox_debug_more("datalen %d, numSamples %lu",dwDataLength, (unsigned long)wav->numSamples);
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
lsx_ima_init_table();
ft->signal.length = wav->numSamples*ft->signal.channels;
break;
case WAVE_FORMAT_GSM610:
wav->numSamples = ((dwDataLength / wav->blockAlign) * wav->samplesPerBlock);
wavgsminit(ft);
ft->signal.length = wav->numSamples*ft->signal.channels;
break;
default:
wav->numSamples = div_bits(dwDataLength, ft->encoding.bits_per_sample) / ft->signal.channels;
ft->signal.length = wav->numSamples * ft->signal.channels;
}
sox_debug("Reading Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wav->formatTag), ft->signal.channels,
wChannels == 1 ? "" : "s", dwSamplesPerSecond);
sox_debug(" %d byte/sec, %d block align, %d bits/samp, %u data bytes",
dwAvgBytesPerSec, wav->blockAlign, wBitsPerSample, dwDataLength);
/* Can also report extended fmt information */
switch (wav->formatTag)
{
case WAVE_FORMAT_ADPCM:
sox_debug(" %d Extsize, %d Samps/block, %lu bytes/block %d Num Coefs, %lu Samps/chan",
wExtSize,wav->samplesPerBlock,(unsigned long)bytesPerBlock,wav->nCoefs,
(unsigned long)wav->numSamples);
break;
case WAVE_FORMAT_IMA_ADPCM:
sox_debug(" %d Extsize, %d Samps/block, %lu bytes/block %lu Samps/chan",
wExtSize, wav->samplesPerBlock, (unsigned long)bytesPerBlock,
(unsigned long)wav->numSamples);
break;
case WAVE_FORMAT_GSM610:
sox_debug("GSM .wav: %d Extsize, %d Samps/block, %lu Samples/chan",
wExtSize, wav->samplesPerBlock, (unsigned long)wav->numSamples);
break;
default:
sox_debug(" %lu Samps/chans", (unsigned long)wav->numSamples);
}
/* Horrible way to find Cool Edit marker points. Taken from Quake source*/
ft->oob.loops[0].start = ~0u;
if(ft->seekable){
/*Got this from the quake source. I think it 32bit aligns the chunks
* doubt any machine writing Cool Edit Chunks writes them at an odd
* offset */
len = (len + 1) & ~1u;
if (lsx_seeki(ft, (ptrdiff_t)len, SEEK_CUR) == SOX_SUCCESS &&
findChunk(ft, "LIST", &len) != SOX_EOF)
{
wav->comment = lsx_malloc((size_t)256);
/* Initialize comment to a NULL string */
wav->comment[0] = 0;
while(!lsx_eof(ft))
{
if (lsx_reads(ft,magic,(size_t)4) == SOX_EOF)
break;
/* First look for type fields for LIST Chunk and
* skip those if found. Since a LIST is a list
* of Chunks, treat the remaining data as Chunks
* again.
*/
if (strncmp(magic, "INFO", (size_t)4) == 0)
{
/*Skip*/
sox_debug("Type INFO");
}
else if (strncmp(magic, "adtl", (size_t)4) == 0)
{
/* Skip */
sox_debug("Type adtl");
}
else
{
if (lsx_readdw(ft,&len) == SOX_EOF)
break;
if (strncmp(magic,"ICRD",(size_t)4) == 0)
{
sox_debug("Chunk ICRD");
if (len > 254)
{
sox_warn("Possible buffer overflow hack attack (ICRD)!");
break;
}
lsx_reads(ft,text, (size_t)len);
if (strlen(wav->comment) + strlen(text) < 254)
{
if (wav->comment[0] != 0)
strcat(wav->comment,"\n");
strcat(wav->comment,text);
}
if (strlen(text) < len)
lsx_seeki(ft, (ptrdiff_t)(len - strlen(text)), SEEK_CUR);
}
else if (strncmp(magic,"ISFT",(size_t)4) == 0)
{
sox_debug("Chunk ISFT");
if (len > 254)
{
sox_warn("Possible buffer overflow hack attack (ISFT)!");
break;
}
lsx_reads(ft,text, (size_t)len);
if (strlen(wav->comment) + strlen(text) < 254)
{
if (wav->comment[0] != 0)
strcat(wav->comment,"\n");
strcat(wav->comment,text);
}
if (strlen(text) < len)
lsx_seeki(ft, (ptrdiff_t)(len - strlen(text)), SEEK_CUR);
}
else if (strncmp(magic,"cue ",(size_t)4) == 0)
{
sox_debug("Chunk cue ");
lsx_seeki(ft,(ptrdiff_t)(len-4),SEEK_CUR);
lsx_readdw(ft,&dwLoopPos);
ft->oob.loops[0].start = dwLoopPos;
}
else if (strncmp(magic,"ltxt",(size_t)4) == 0)
{
sox_debug("Chunk ltxt");
lsx_readdw(ft,&dwLoopPos);
ft->oob.loops[0].length = dwLoopPos - ft->oob.loops[0].start;
if (len > 4)
lsx_seeki(ft, (ptrdiff_t)(len - 4), SEEK_CUR);
}
else
{
sox_debug("Attempting to seek beyond unsupported chunk '%c%c%c%c' of length %d bytes", magic[0], magic[1], magic[2], magic[3], len);
len = (len + 1) & ~1u;
lsx_seeki(ft, (ptrdiff_t)len, SEEK_CUR);
}
}
}
}
lsx_clearerr(ft);
lsx_seeki(ft,(ptrdiff_t)wav->dataStart,SEEK_SET);
}
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 = (priv_t *) ft->priv;
size_t done;
ft->sox_errno = SOX_SUCCESS;
/* 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:
if (!wav->ignoreSize && len > (wav->numSamples*ft->signal.channels))
len = (wav->numSamples*ft->signal.channels);
done = 0;
while (done < len) { /* Still want data? */
/* 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 */
{
short *p, *top;
size_t ct;
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;
break;
case SOX_ENCODING_GSM:
if (!wav->ignoreSize && len > wav->numSamples*ft->signal.channels)
len = (wav->numSamples*ft->signal.channels);
done = wavgsmread(ft, buf, len);
if (done == 0 && wav->numSamples != 0)
sox_warn("Premature EOF on .wav input file");
break;
default: /* assume PCM or float encoding */
if (!wav->ignoreSize && len > wav->numSamples*ft->signal.channels)
len = (wav->numSamples*ft->signal.channels);
done = lsx_rawread(ft, buf, len);
/* If software thinks there are more samples but I/O */
/* says otherwise, let the user know about this. */
if (done == 0 && wav->numSamples != 0)
sox_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->comment);
wav->comment = NULL;
switch (ft->encoding.encoding)
{
case SOX_ENCODING_GSM:
wavgsmdestroy(ft);
break;
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)
sox_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;
case WAVE_FORMAT_GSM610:
return wavgsminit(ft);
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
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
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
(+ a padding byte if dwDataLength is odd)
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
*/
#define MS_UNSPEC 0x7ffff000 /* Unspecified data size (this is a kludge) */
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 */
uint32_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 */
uint16_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 */
uint32_t dwSamplesWritten=0; /* windows doesnt seem to use this*/
/* data chunk */
uint32_t dwDataLength = MS_UNSPEC; /* 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) */
long blocksWritten = 0;
sox_bool isExtensible = sox_false; /* WAVE_FORMAT_EXTENSIBLE? */
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;
case SOX_ENCODING_GSM:
if (wChannels!=1)
{
sox_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;
default:
break;
}
wav->formatTag = wFormatTag;
wav->blockAlign = wBlockAlign;
wav->samplesPerBlock = wSamplesPerBlock;
if (!second_header && !ft->signal.length) { /* adjust for blockAlign */
blocksWritten = dwDataLength/wBlockAlign;
dwDataLength = blocksWritten * wBlockAlign;
dwSamplesWritten = blocksWritten * wSamplesPerBlock;
} else { /* fixup with real length */
dwSamplesWritten = second_header? wav->numSamples : ft->signal.length;
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 && strcmp(ft->filetype, "wavpcm")) || wChannels > 2)
{
isExtensible = sox_true;
wFmtSize += 2 + 22;
}
else if (wFormatTag != WAVE_FORMAT_PCM)
wFmtSize += 2+wExtSize; /* plus ExtData */
wRiffLength = 4 + (8+wFmtSize) + (8+dwDataLength);
if (wFormatTag != WAVE_FORMAT_PCM) /* PCM omits the "fact" chunk */
wRiffLength += (8+dwFactSize);
/* 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)
sox_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) {
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 */
lsx_writew(ft, 22);
lsx_writew(ft, wBitsPerSample); /* No padding in container */
lsx_writedw(ft, 0); /* Speaker mapping not specified */
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) {
sox_debug("Writing Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wFormatTag), wChannels,
wChannels == 1 ? "" : "s", dwSamplesPerSecond);
sox_debug(" %d byte/sec, %d block align, %d bits/samp",
dwAvgBytesPerSec, wBlockAlign, wBitsPerSample);
} else {
sox_debug("Finished writing Wave file, %u data bytes %lu samples",
dwDataLength,(unsigned long)wav->numSamples);
if (wFormatTag == WAVE_FORMAT_GSM610){
sox_debug("GSM6.10 format: %li blocks %u padded samples %u padded data bytes",
blocksWritten, dwSamplesWritten, dwDataLength);
if (wav->gsmbytecount != dwDataLength)
sox_warn("help ! internal inconsistency - data_written %u gsmbytecount %lu",
dwDataLength, (unsigned long)wav->gsmbytecount);
}
}
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;
case WAVE_FORMAT_GSM610:
len = wavgsmwrite(ft, buf, len);
wav->numSamples += (len/ft->signal.channels);
return len;
break;
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;
case WAVE_FORMAT_GSM610:
wavgsmstopwrite(ft);
break;
}
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 == ft->signal.length)
return SOX_SUCCESS;
if (!ft->seekable)
return SOX_EOF;
if (lsx_seeki(ft, (ptrdiff_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 char *wav_format_str(unsigned wFormatTag)
{
switch (wFormatTag)
{
case WAVE_FORMAT_UNKNOWN:
return "Microsoft Official Unknown";
case WAVE_FORMAT_PCM:
return "Microsoft PCM";
case WAVE_FORMAT_ADPCM:
return "Microsoft ADPCM";
case WAVE_FORMAT_IEEE_FLOAT:
return "IEEE Float";
case WAVE_FORMAT_ALAW:
return "Microsoft A-law";
case WAVE_FORMAT_MULAW:
return "Microsoft U-law";
case WAVE_FORMAT_OKI_ADPCM:
return "OKI ADPCM format.";
case WAVE_FORMAT_IMA_ADPCM:
return "IMA ADPCM";
case WAVE_FORMAT_DIGISTD:
return "Digistd format.";
case WAVE_FORMAT_DIGIFIX:
return "Digifix format.";
case WAVE_FORMAT_DOLBY_AC2:
return "Dolby AC2";
case WAVE_FORMAT_GSM610:
return "GSM 6.10";
case WAVE_FORMAT_ROCKWELL_ADPCM:
return "Rockwell ADPCM";
case WAVE_FORMAT_ROCKWELL_DIGITALK:
return "Rockwell DIGITALK";
case WAVE_FORMAT_G721_ADPCM:
return "G.721 ADPCM";
case WAVE_FORMAT_G728_CELP:
return "G.728 CELP";
case WAVE_FORMAT_MPEG:
return "MPEG";
case WAVE_FORMAT_MPEGLAYER3:
return "MPEG Layer 3";
case WAVE_FORMAT_G726_ADPCM:
return "G.726 ADPCM";
case WAVE_FORMAT_G722_ADPCM:
return "G.722 ADPCM";
default:
return "Unknown";
}
}
static int seek(sox_format_t * ft, size_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 new_offset, 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, (ptrdiff_t)(gsmoff + wav->dataStart), SEEK_SET);
if (ft->sox_errno == SOX_SUCCESS) {
/* offset is in samples */
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, (ptrdiff_t)wav->dataStart + (ptrdiff_t)to, SEEK_SET);
if (ft->sox_errno == SOX_SUCCESS)
wav->numSamples -= (size_t)wide_sample / ft->signal.channels;
}
return ft->sox_errno;
}
SOX_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,
SOX_ENCODING_GSM, 0,
SOX_ENCODING_MS_ADPCM, 4, 0,
SOX_ENCODING_IMA_ADPCM, 4, 0,
SOX_ENCODING_FLOAT, 32, 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;
}