ref: 03a694872ae017ecf5aaa62592626a7545197937
dir: /src/wav.c/
/*
* Microsoft's WAVE sound format driver
*
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Lance Norskog And Sundry Contributors are not responsible for
* the consequences of using this software.
*
* Change History:
*
* November 23, 1999 - Stan Brooks (stabro@megsinet.com)
* Merged in gsm support patches from Stuart Daines...
* Since we had simultaneously made similar changes in
* wavwriteheader() and wavstartread(), this was some
* work. Hopefully the result is cleaner than either
* version, and nothing broke.
*
* November 20, 1999 - Stan Brooks (stabro@megsinet.com)
* Mods for faster adpcm decoding and addition of IMA_ADPCM
* and ADPCM writing... low-level codex functions moved to
* external modules ima_rw.c and adpcm.c. Some general cleanup,
* consistent with writing adpcm and other output formats.
* Headers written for adpcm include the 'fact' subchunk.
*
* September 11, 1998 - Chris Bagwell (cbagwell@sprynet.com)
* Fixed length bug for IMA and MS ADPCM files.
*
* June 1, 1998 - Chris Bagwell (cbagwell@sprynet.com)
* Fixed some compiler warnings as reported by Kjetil Torgrim Homme
* <kjetilho@ifi.uio.no>.
* Fixed bug that caused crashes when reading mono MS ADPCM files. Patch
* was sent from Michael Brown (mjb@pootle.demon.co.uk).
*
* March 15, 1998 - Chris Bagwell (cbagwell@sprynet.com)
* Added support for Microsoft's ADPCM and IMA (or better known as
* DVI) ADPCM format for wav files. Thanks goes to Mark Podlipec's
* XAnim code. It gave some real life understanding of how the ADPCM
* format is processed. Actual code was implemented based off of
* various sources from the net.
*
* NOTE: Previous maintainers weren't very good at providing contact
* information.
*
* Copyright 1992 Rick Richardson
* Copyright 1991 Lance Norskog And Sundry Contributors
*
* Fixed by various contributors previous to 1998:
* 1) Little-endian handling
* 2) Skip other kinds of file data
* 3) Handle 16-bit formats correctly
* 4) Not go into infinite loop
*
* User options should override file header - we assumed user knows what
* they are doing if they specify options.
* Enhancements and clean up by Graeme W. Gill, 93/5/17
*
* Info for format tags can be found at:
* http://www.microsoft.com/asf/resources/draft-ietf-fleischman-codec-subtree-01.txt
*
*/
#include <string.h> /* Included for strncmp */
#include <stdlib.h> /* Included for malloc and free */
#include <stdio.h>
#ifdef HAVE_UNISTD_H
#include <unistd.h> /* For SEEK_* defines if not found in stdio */
#endif
#include "st.h"
#include "wav.h"
#include "ima_rw.h"
#include "adpcm.h"
#ifdef HAVE_LIBGSM
#include "gsm.h"
#endif
#undef PAD_NSAMPS
/* #define PAD_NSAMPS */
/* Private data for .wav file */
typedef struct wavstuff {
LONG numSamples; /* reading: starts at total count and decremented */
/* writing: starts at 0 and counts samples written */
LONG dataLength; /* needed for ADPCM writing */
unsigned short formatTag; /* What type of encoding file is using */
unsigned short samplesPerBlock;
unsigned short blockAlign;
/* following used by *ADPCM wav files */
unsigned short nCoefs; /* ADPCM: number of coef sets */
short *iCoefs; /* 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 */
#ifdef HAVE_LIBGSM
gsm gsmhandle;
gsm_signal *gsmsample;
int gsmindex;
int gsmbytecount; /* counts bytes written to data block */
#endif
} *wav_t;
/*
#if sizeof(struct wavstuff) > PRIVSIZE
# warn "Uh-Oh"
#endif
*/
static char *wav_format_str();
static int wavwritehdr(P2(ft_t, int));
/****************************************************************************/
/* IMA ADPCM Support Functions Section */
/****************************************************************************/
/*
*
* ImaAdpcmReadBlock - Grab and decode complete block of samples
*
*/
unsigned short ImaAdpcmReadBlock(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
int bytesRead;
int samplesThisBlock;
/* Pull in the packet and check the header */
bytesRead = fread(wav->packet,1,wav->blockAlign,ft->fp);
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 = ImaSamplesIn(0, ft->info.channels, bytesRead, 0);
if (samplesThisBlock == 0)
{
st_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 */
ImaBlockExpandI(ft->info.channels, wav->packet, wav->samples, samplesThisBlock);
return samplesThisBlock;
}
/****************************************************************************/
/* MS ADPCM Support Functions Section */
/****************************************************************************/
/*
*
* AdpcmReadBlock - Grab and decode complete block of samples
*
*/
unsigned short AdpcmReadBlock(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
int bytesRead;
int samplesThisBlock;
const char *errmsg;
/* Pull in the packet and check the header */
bytesRead = fread(wav->packet,1,wav->blockAlign,ft->fp);
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 = AdpcmSamplesIn(0, ft->info.channels, bytesRead, 0);
if (samplesThisBlock == 0)
{
st_warn("Premature EOF on .wav input file");
return 0;
}
}
errmsg = AdpcmBlockExpandI(ft->info.channels, wav->nCoefs, wav->iCoefs, wav->packet, wav->samples, samplesThisBlock);
if (errmsg)
st_warn((char*)errmsg);
return samplesThisBlock;
}
/****************************************************************************/
/* Common ADPCM Write Function */
/****************************************************************************/
static int xxxAdpcmWriteBlock(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
int chans, ct;
short *p;
chans = ft->info.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) {
AdpcmBlockMashI(chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, wav->blockAlign,9);
}else{ /* WAVE_FORMAT_IMA_ADPCM */
ImaBlockMashI(chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, 9);
}
/* write the compressed packet */
if (fwrite(wav->packet, wav->blockAlign, 1, ft->fp) != 1)
{
st_fail("write error");
return (ST_EOF);
}
/* update lengths and samplePtr */
wav->dataLength += wav->blockAlign;
#ifndef PAD_NSAMPS
wav->numSamples += ct/chans;
#else
wav->numSamples += wav->samplesPerBlock;
#endif
wav->samplePtr = wav->samples;
}
return (ST_SUCCESS);
}
/****************************************************************************/
/* WAV GSM6.10 support functions */
/****************************************************************************/
#ifdef HAVE_LIBGSM
/* create the gsm object, malloc buffer for 160*2 samples */
int wavgsminit(ft)
ft_t ft;
{
int valueP=1;
wav_t wav = (wav_t) ft->priv;
wav->gsmbytecount=0;
wav->gsmhandle=gsm_create();
if (!wav->gsmhandle)
{
st_fail("cannot create GSM object");
return (ST_EOF);
}
if(gsm_option(wav->gsmhandle,GSM_OPT_WAV49,&valueP) == -1){
st_fail("error setting gsm_option for WAV49 format. Recompile gsm library with -DWAV49 option and relink sox");
return (ST_EOF);
}
wav->gsmsample=malloc(sizeof(gsm_signal)*160*2);
if (wav->gsmsample == NULL){
st_fail("error allocating memory for gsm buffer");
return (ST_EOF);
}
wav->gsmindex=0;
return (ST_SUCCESS);
}
/*destroy the gsm object and free the buffer */
void wavgsmdestroy(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
gsm_destroy(wav->gsmhandle);
free(wav->gsmsample);
}
LONG wavgsmread(ft, buf, len)
ft_t ft;
LONG *buf, len;
{
wav_t wav = (wav_t) ft->priv;
int done=0;
int bytes;
gsm_byte frame[65];
/* copy out any samples left from the last call */
while(wav->gsmindex && (wav->gsmindex<160*2) && (done < len))
buf[done++]=LEFT(wav->gsmsample[wav->gsmindex++],16);
/* read and decode loop, possibly leaving some samples in wav->gsmsample */
while (done < len) {
wav->gsmindex=0;
bytes = fread(frame,1,65,ft->fp);
if (bytes <=0)
return done;
if (bytes<65) {
st_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)
{
st_fail("error during gsm decode");
return 0;
}
/* decode the short 32 byte half */
if(gsm_decode(wav->gsmhandle,frame+33, wav->gsmsample+160)<0)
{
st_fail("error during gsm decode");
return 0;
}
while ((wav->gsmindex <160*2) && (done < len)){
buf[done++]=LEFT(wav->gsmsample[(wav->gsmindex)++],16);
}
}
return done;
}
static int wavgsmflush(ft, pad)
ft_t ft;
int pad; /* normally 0, but 1 to pad last write to even datalen */
{
gsm_byte frame[65];
wav_t wav = (wav_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 (fwrite(frame, 1, 65, ft->fp) != 65)
{
st_fail("write error");
return (ST_EOF);
}
wav->gsmbytecount += 65;
wav->gsmindex = 0;
if (pad & wav->gsmbytecount){
/* pad output to an even number of bytes */
if(st_writeb(ft, 0))
{
st_fail("write error");
return (ST_EOF);
}
wav->gsmbytecount += 1;
}
return (ST_SUCCESS);
}
LONG wavgsmwrite(ft, buf, len)
ft_t ft;
LONG *buf, len;
{
wav_t wav = (wav_t) ft->priv;
int done = 0;
int rc;
while (done < len) {
while ((wav->gsmindex < 160*2) && (done < len))
wav->gsmsample[(wav->gsmindex)++] = RIGHT(buf[done++], 16);
if (wav->gsmindex < 160*2)
break;
rc = wavgsmflush(ft, 0);
if (rc)
return 0;
}
return done;
}
void wavgsmstopwrite(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
if (wav->gsmindex)
wavgsmflush(ft, 1);
wavgsmdestroy(ft);
}
#endif /*ifdef out gsm code */
/****************************************************************************/
/* General Sox WAV file code */
/****************************************************************************/
/* FIXME: Use common misc.c skip code */
static void fSkip(FILE *fp, ULONG len)
{
while (len > 0 && !feof(fp))
{
getc(fp);
len--;
}
}
static ULONG findChunk(ft_t ft, const char *Label)
{
char magic[5];
ULONG len;
for (;;)
{
if (st_reads(ft, magic, 4) == ST_EOF)
{
st_fail("WAVE file has missing %s chunk", Label);
return 0;
}
st_readdw(ft, &len);
if (strncmp(Label, magic, 4) == 0)
break; /* Found the data chunk */
fSkip(ft->fp, len); /* skip to next chunk */
}
return len;
}
/*
* Do anything required before you start reading samples.
* Read file header.
* Find out sampling rate,
* size and encoding of samples,
* mono/stereo/quad.
*/
int st_wavstartread(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
char magic[5];
ULONG len;
int rc;
/* wave file characteristics */
ULONG wRiffLength;
unsigned short wChannels; /* number of channels */
ULONG wSamplesPerSecond; /* samples per second per channel */
ULONG wAvgBytesPerSec; /* estimate of bytes per second needed */
unsigned short wBitsPerSample; /* bits per sample */
unsigned short wFmtSize;
unsigned short wExtSize = 0; /* extended field for non-PCM */
ULONG wDataLength; /* length of sound data in bytes */
ULONG bytesPerBlock = 0;
ULONG bytespersample; /* bytes per sample (per channel */
if (ST_IS_BIGENDIAN) ft->swap = ft->swap ? 0 : 1;
if (st_reads(ft, magic, 4) == ST_EOF || strncmp("RIFF", magic, 4))
{
st_fail("WAVE: RIFF header not found");
return ST_EOF;
}
st_readdw(ft, &wRiffLength);
if (st_reads(ft, magic, 4) == ST_EOF || strncmp("WAVE", magic, 4))
{
st_fail("WAVE header not found");
return ST_EOF;
}
/* Now look for the format chunk */
wFmtSize = len = findChunk(ft, "fmt ");
/* findChunk() only returns if chunk was found */
if (wFmtSize < 16)
{
st_fail("WAVE file fmt chunk is too short");
return ST_EOF;
}
st_readw(ft, &(wav->formatTag));
st_readw(ft, &wChannels);
st_readdw(ft, &wSamplesPerSecond);
st_readdw(ft, &wAvgBytesPerSec); /* Average bytes/second */
st_readw(ft, &(wav->blockAlign)); /* Block align */
st_readw(ft, &wBitsPerSample); /* bits per sample per channel */
len -= 16;
switch (wav->formatTag)
{
case WAVE_FORMAT_UNKNOWN:
st_fail("WAVE file is in unsupported Microsoft Official Unknown format.");
return ST_EOF;
case WAVE_FORMAT_PCM:
/* Default (-1) depends on sample size. Set that later on. */
if (ft->info.encoding != -1 && ft->info.encoding != ST_ENCODING_UNSIGNED &&
ft->info.encoding != ST_ENCODING_SIGN2)
st_warn("User options overriding encoding read in .wav header");
/* Needed by rawread() functions */
rc = st_rawstartread(ft);
if (rc)
return rc;
break;
case WAVE_FORMAT_IMA_ADPCM:
if (ft->info.encoding == -1 || ft->info.encoding == ST_ENCODING_IMA_ADPCM)
ft->info.encoding = ST_ENCODING_IMA_ADPCM;
else
st_warn("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_ADPCM:
if (ft->info.encoding == -1 || ft->info.encoding == ST_ENCODING_ADPCM)
ft->info.encoding = ST_ENCODING_ADPCM;
else
st_warn("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_IEEE_FLOAT:
st_fail("Sorry, this WAV file is in IEEE Float format.");
return ST_EOF;
case WAVE_FORMAT_ALAW:
if (ft->info.encoding == -1 || ft->info.encoding == ST_ENCODING_ALAW)
ft->info.encoding = ST_ENCODING_ALAW;
else
st_warn("User options overriding encoding read in .wav header");
/* Needed by rawread() functions */
rc = st_rawstartread(ft);
if (rc)
return rc;
break;
case WAVE_FORMAT_MULAW:
if (ft->info.encoding == -1 || ft->info.encoding == ST_ENCODING_ULAW)
ft->info.encoding = ST_ENCODING_ULAW;
else
st_warn("User options overriding encoding read in .wav header");
/* Needed by rawread() functions */
rc = st_rawstartread(ft);
if (rc)
return rc;
break;
case WAVE_FORMAT_OKI_ADPCM:
st_fail("Sorry, this WAV file is in OKI ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_DIGISTD:
st_fail("Sorry, this WAV file is in Digistd format.");
return ST_EOF;
case WAVE_FORMAT_DIGIFIX:
st_fail("Sorry, this WAV file is in Digifix format.");
return ST_EOF;
case WAVE_FORMAT_DOLBY_AC2:
st_fail("Sorry, this WAV file is in Dolby AC2 format.");
return ST_EOF;
case WAVE_FORMAT_GSM610:
#ifdef HAVE_LIBGSM
if (ft->info.encoding == -1 || ft->info.encoding == ST_ENCODING_GSM )
ft->info.encoding = ST_ENCODING_GSM;
else
st_warn("User options overriding encoding read in .wav header");
break;
#else
st_fail("Sorry, this WAV file is in GSM6.10 format and no GSM support present, recompile sox with gsm library");
return ST_EOF;
#endif
case WAVE_FORMAT_ROCKWELL_ADPCM:
st_fail("Sorry, this WAV file is in Rockwell ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_ROCKWELL_DIGITALK:
st_fail("Sorry, this WAV file is in Rockwell DIGITALK format.");
return ST_EOF;
case WAVE_FORMAT_G721_ADPCM:
st_fail("Sorry, this WAV file is in G.721 ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_G728_CELP:
st_fail("Sorry, this WAV file is in G.728 CELP format.");
return ST_EOF;
case WAVE_FORMAT_MPEG:
st_fail("Sorry, this WAV file is in MPEG format.");
return ST_EOF;
case WAVE_FORMAT_MPEGLAYER3:
st_fail("Sorry, this WAV file is in MPEG Layer 3 format.");
return ST_EOF;
case WAVE_FORMAT_G726_ADPCM:
st_fail("Sorry, this WAV file is in G.726 ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_G722_ADPCM:
st_fail("Sorry, this WAV file is in G.722 ADPCM format.");
return ST_EOF;
default: st_fail("WAV file has unknown format type of %x",wav->formatTag);
return ST_EOF;
}
/* User options take precedence */
if (ft->info.channels == -1 || ft->info.channels == wChannels)
ft->info.channels = wChannels;
else
st_warn("User options overriding channels read in .wav header");
if (ft->info.rate == 0 || ft->info.rate == wSamplesPerSecond)
ft->info.rate = wSamplesPerSecond;
else
st_warn("User options overriding rate read in .wav header");
wav->iCoefs = NULL;
wav->packet = NULL;
wav->samples = NULL;
/* non-PCM formats have extended fmt chunk. Check for those cases. */
if (wav->formatTag != WAVE_FORMAT_PCM) {
if (len >= 2) {
st_readw(ft, &wExtSize);
len -= 2;
} else {
st_warn("wave header missing FmtExt chunk");
}
}
if (wExtSize > len)
{
st_fail("wave header error: wExtSize inconsistent with wFmtLen");
return ST_EOF;
}
switch (wav->formatTag)
{
/* ULONG max_spb; */
case WAVE_FORMAT_ADPCM:
if (wExtSize < 4)
{
st_fail("format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 4);
return ST_EOF;
}
if (wBitsPerSample != 4)
{
st_fail("Can only handle 4-bit MS ADPCM in wav files");
return ST_EOF;
}
st_readw(ft, &(wav->samplesPerBlock));
bytesPerBlock = AdpcmBytesPerBlock(ft->info.channels, wav->samplesPerBlock);
if (bytesPerBlock > wav->blockAlign)
{
st_fail("format[%s]: samplesPerBlock(%d) incompatible with blockAlign(%d)",
wav_format_str(wav->formatTag), wav->samplesPerBlock, wav->blockAlign);
return ST_EOF;
}
st_readw(ft, &(wav->nCoefs));
if (wav->nCoefs < 7 || wav->nCoefs > 0x100) {
st_fail("ADPCM file nCoefs (%.4hx) makes no sense\n", wav->nCoefs);
return ST_EOF;
}
wav->packet = (unsigned char *)malloc(wav->blockAlign);
if (!wav->packet)
{
st_fail("Unable to alloc resources");
return ST_EOF;
}
len -= 4;
if (wExtSize < 4 + 4*wav->nCoefs)
{
st_fail("wave header error: wExtSize(%d) too small for nCoefs(%d)", wExtSize, wav->nCoefs);
return ST_EOF;
}
wav->samples = (short *)malloc(wChannels*wav->samplesPerBlock*sizeof(short));
if (!wav->samples)
{
st_fail("Unable to alloc resources");
return ST_EOF;
}
/* nCoefs, iCoefs used by adpcm.c */
wav->iCoefs = (short *)malloc(wav->nCoefs * 2 * sizeof(short));
if (!wav->iCoefs)
{
st_fail("Unable to alloc resources");
return ST_EOF;
}
{
int i, errct=0;
for (i=0; len>=2 && i < 2*wav->nCoefs; i++) {
st_readw(ft, &(wav->iCoefs[i]));
len -= 2;
if (i<14) errct += (wav->iCoefs[i] != iCoef[i/2][i%2]);
/* fprintf(stderr,"iCoefs[%2d] %4d\n",i,wav->iCoefs[i]); */
}
if (errct) st_warn("base iCoefs differ in %d/14 positions",errct);
}
bytespersample = ST_SIZE_WORD; /* AFTER de-compression */
break;
case WAVE_FORMAT_IMA_ADPCM:
if (wExtSize < 2)
{
st_fail("format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 2);
return ST_EOF;
}
if (wBitsPerSample != 4)
{
st_fail("Can only handle 4-bit IMA ADPCM in wav files");
return ST_EOF;
}
st_readw(ft, &(wav->samplesPerBlock));
bytesPerBlock = ImaBytesPerBlock(ft->info.channels, wav->samplesPerBlock);
if (bytesPerBlock > wav->blockAlign || wav->samplesPerBlock%8 != 1)
{
st_fail("format[%s]: samplesPerBlock(%d) incompatible with blockAlign(%d)",
wav_format_str(wav->formatTag), wav->samplesPerBlock, wav->blockAlign);
return ST_EOF;
}
wav->packet = (unsigned char *)malloc(wav->blockAlign);
if (!wav->packet)
{
st_fail("Unable to alloc resources");
return ST_EOF;
}
len -= 2;
wav->samples = (short *)malloc(wChannels*wav->samplesPerBlock*sizeof(short));
if (!wav->samples)
{
st_fail("Unable to alloc resources");
return ST_EOF;
}
bytespersample = ST_SIZE_WORD; /* AFTER de-compression */
break;
#ifdef HAVE_LIBGSM
/* GSM formats have extended fmt chunk. Check for those cases. */
case WAVE_FORMAT_GSM610:
if (wExtSize < 2)
{
st_fail("format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 2);
return ST_EOF;
}
st_readw(ft, &wav->samplesPerBlock);
bytesPerBlock = 65;
if (wav->blockAlign != 65)
{
st_fail("format[%s]: expects blockAlign(%d) = %d",
wav_format_str(wav->formatTag), wav->blockAlign, 65);
return ST_EOF;
}
if (wav->samplesPerBlock != 320)
{
st_fail("format[%s]: expects samplesPerBlock(%d) = %d",
wav_format_str(wav->formatTag), wav->samplesPerBlock, 320);
return ST_EOF;
}
bytespersample = ST_SIZE_WORD; /* AFTER de-compression */
len -= 2;
break;
#endif
default:
bytespersample = (wBitsPerSample + 7)/8;
}
switch (bytespersample)
{
case ST_SIZE_BYTE:
/* User options take precedence */
if (ft->info.size == -1 || ft->info.size == ST_SIZE_BYTE)
ft->info.size = ST_SIZE_BYTE;
else
st_warn("User options overriding size read in .wav header");
/* Now we have enough information to set default encodings. */
if (ft->info.encoding == -1)
ft->info.encoding = ST_ENCODING_UNSIGNED;
break;
case ST_SIZE_WORD:
if (ft->info.size == -1 || ft->info.size == ST_SIZE_WORD)
ft->info.size = ST_SIZE_WORD;
else
st_warn("User options overriding size read in .wav header");
/* Now we have enough information to set default encodings. */
if (ft->info.encoding == -1)
ft->info.encoding = ST_ENCODING_SIGN2;
break;
case ST_SIZE_DWORD:
if (ft->info.size == -1 || ft->info.size == ST_SIZE_DWORD)
ft->info.size = ST_SIZE_DWORD;
else
st_warn("User options overriding size read in .wav header");
/* Now we have enough information to set default encodings. */
if (ft->info.encoding == -1)
ft->info.encoding = ST_ENCODING_SIGN2;
break;
default:
st_fail("Sorry, don't understand .wav size");
return ST_EOF;
}
/* Skip anything left over from fmt chunk */
fSkip(ft->fp, len);
/* for non-PCM formats, there's a 'fact' chunk before
* the upcoming 'data' chunk */
/* Now look for the wave data chunk */
wDataLength = len = findChunk(ft, "data");
/* findChunk() only returns if chunk was found */
switch (wav->formatTag)
{
case WAVE_FORMAT_ADPCM:
wav->numSamples =
AdpcmSamplesIn(wDataLength, ft->info.channels, wav->blockAlign, wav->samplesPerBlock);
/*st_report("datalen %d, numSamples %d",wDataLength, wav->numSamples);*/
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 =
ImaSamplesIn(wDataLength, ft->info.channels, wav->blockAlign, wav->samplesPerBlock);
/*st_report("datalen %d, numSamples %d",wDataLength, wav->numSamples);*/
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
initImaTable();
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
wav->numSamples = (((wDataLength / wav->blockAlign) * wav->samplesPerBlock) * ft->info.channels);
wavgsminit(ft);
break;
#endif
default:
wav->numSamples = wDataLength/ft->info.size; /* total samples */
}
st_report("Reading Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wav->formatTag), ft->info.channels,
wChannels == 1 ? "" : "s", wSamplesPerSecond);
st_report(" %d byte/sec, %d block align, %d bits/samp, %u data bytes",
wAvgBytesPerSec, wav->blockAlign, wBitsPerSample, wDataLength);
/* Can also report extended fmt information */
switch (wav->formatTag)
{
case WAVE_FORMAT_ADPCM:
st_report(" %d Extsize, %d Samps/block, %d bytes/block %d Num Coefs",
wExtSize,wav->samplesPerBlock,bytesPerBlock,wav->nCoefs);
break;
case WAVE_FORMAT_IMA_ADPCM:
st_report(" %d Extsize, %d Samps/block, %d bytes/block",
wExtSize,wav->samplesPerBlock,bytesPerBlock);
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
st_report("GSM .wav: %d Extsize, %d Samps/block, %d samples",
wExtSize,wav->samplesPerBlock,wav->numSamples);
break;
#endif
default:
break;
}
return ST_SUCCESS;
}
/*
* Read up to len samples from file.
* Convert to signed longs.
* Place in buf[].
* Return number of samples read.
*/
LONG st_wavread(ft, buf, len)
ft_t ft;
LONG *buf, len;
{
wav_t wav = (wav_t) ft->priv;
LONG done;
/* If file is in ADPCM encoding then read in multiple blocks else */
/* read as much as possible and return quickly. */
switch (ft->info.encoding)
{
case ST_ENCODING_IMA_ADPCM:
case ST_ENCODING_ADPCM:
/* FIXME: numSamples is not used consistently in
* wav handler. Sometimes it accounts for stereo,
* sometimes it does not.
*/
if (len > (wav->numSamples*ft->info.channels))
len = (wav->numSamples*ft->info.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 short to LONG */
{
short *p, *top;
int ct;
ct = len-done;
if (ct > (wav->blockSamplesRemaining*ft->info.channels))
ct = (wav->blockSamplesRemaining*ft->info.channels);
done += ct;
wav->blockSamplesRemaining -= (ct/ft->info.channels);
p = wav->samplePtr;
top = p+ct;
/* Output is already signed */
while (p<top)
*buf++ = LEFT((*p++), 16);
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->info.channels);
return done;
break;
#ifdef HAVE_LIBGSM
case ST_ENCODING_GSM:
if (len > wav->numSamples)
len = wav->numSamples;
done = wavgsmread(ft, buf, len);
if (done == 0 && wav->numSamples != 0)
st_warn("Premature EOF on .wav input file");
break;
#endif
default: /* assume PCM encoding */
if (len > wav->numSamples)
len = wav->numSamples;
done = st_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)
st_warn("Premature EOF on .wav input file");
}
wav->numSamples -= done;
return done;
}
/*
* Do anything required when you stop reading samples.
* Don't close input file!
*/
int st_wavstopread(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
int rc = ST_SUCCESS;
if (wav->packet) free(wav->packet);
if (wav->samples) free(wav->samples);
if (wav->iCoefs) free(wav->iCoefs);
switch (ft->info.encoding)
{
#ifdef HAVE_LIBGSM
case ST_ENCODING_GSM:
wavgsmdestroy(ft);
break;
#endif
case ST_ENCODING_IMA_ADPCM:
case ST_ENCODING_ADPCM:
break;
default:
/* Needed for rawread() */
rc = st_rawstopread(ft);
}
return rc;
}
int st_wavstartwrite(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
int rc;
if (ST_IS_BIGENDIAN) ft->swap = ft->swap ? 0 : 1;
wav->numSamples = 0;
wav->dataLength = 0;
if (!ft->seekable)
st_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->iCoefs = NULL;
switch (wav->formatTag)
{
int ch, sbsize;
case WAVE_FORMAT_IMA_ADPCM:
initImaTable();
/* intentional case fallthru! */
case WAVE_FORMAT_ADPCM:
/* #channels already range-checked for overflow in wavwritehdr() */
for (ch=0; ch<ft->info.channels; ch++)
wav->state[ch] = 0;
sbsize = ft->info.channels * wav->samplesPerBlock;
wav->packet = (unsigned char *)malloc(wav->blockAlign);
wav->samples = (short *)malloc(sbsize*sizeof(short));
if (!wav->packet || !wav->samples)
{
st_fail("Unable to alloc resources");
return ST_EOF;
}
wav->sampleTop = wav->samples + sbsize;
wav->samplePtr = wav->samples;
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
wavgsminit(ft);
break;
#endif
default:
break;
}
return ST_SUCCESS;
}
/* wavwritehdr: write .wav headers as follows:
bytes variable description
0 - 3 'RIFF'
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 wSamplesPerSecond samples per second per channel
28 - 31 wAvgBytesPerSec 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 wDataLength length of data chunk minus 8 byte header
44 - (wDataLength + 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 wFactSize = 4 length of the fact chunk minus 8 byte header
46 - 49 wSamplesWritten actual number of samples written out
50 - 53 'data'
54 - 57 wDataLength length of data chunk minus 8 byte header
58 - (wDataLength + 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 wFactSize = 4 length of the fact chunk minus 8 byte header
48 - 51 wSamplesWritten actual number of samples written out
52 - 55 'data'
56 - 59 wDataLength length of data chunk minus 8 byte header
60 - (wDataLength + 59) the data
(+ a padding byte if wDataLength 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
wSamplesWritten - (fact header) the number of samples written (after padding
to a complete block eg for GSM)
wDataLength - (data chunk header) the number of (valid) data bytes written
*/
static int wavwritehdr(ft, second_header)
ft_t ft;
int second_header;
{
wav_t wav = (wav_t) ft->priv;
/* variables written to wav file header */
/* RIFF header */
ULONG wRiffLength ; /* length of file after 8 byte riff header */
/* fmt chunk */
ULONG wFmtSize = 16; /* size field of the fmt chunk */
unsigned short wFormatTag = 0; /* data format */
unsigned short wChannels; /* number of channels */
ULONG wSamplesPerSecond; /* samples per second per channel*/
ULONG wAvgBytesPerSec=0; /* estimate of bytes per second needed */
unsigned short wBlockAlign=0; /* byte alignment of a basic sample block */
unsigned short wBitsPerSample=0; /* bits per sample */
/* fmt chunk extension (not PCM) */
unsigned short wExtSize=0; /* extra bytes in the format extension */
unsigned short wSamplesPerBlock; /* samples per channel per block */
/* wSamplesPerBlock and other things may go into format extension */
/* fact chunk (not PCM) */
ULONG wFactSize=4; /* length of the fact chunk */
ULONG wSamplesWritten=0; /* windows doesnt seem to use this*/
/* data chunk */
ULONG wDataLength=0x7ffff000L; /* length of sound data in bytes */
/* end of variables written to header */
/* internal variables, intermediate values etc */
ULONG bytespersample; /* (uncompressed) bytes per sample (per channel) */
ULONG blocksWritten = 0;
int rc;
wSamplesPerSecond = ft->info.rate;
wChannels = ft->info.channels;
switch (ft->info.size)
{
case ST_SIZE_BYTE:
wBitsPerSample = 8;
if (ft->info.encoding != ST_ENCODING_UNSIGNED &&
ft->info.encoding != ST_ENCODING_ULAW &&
ft->info.encoding != ST_ENCODING_ALAW &&
ft->info.encoding != ST_ENCODING_GSM)
{
st_warn("Do not support %s with 8-bit data. Forcing to unsigned",st_encodings_str[ft->info.encoding]);
ft->info.encoding = ST_ENCODING_UNSIGNED;
}
break;
case ST_SIZE_WORD:
wBitsPerSample = 16;
if (ft->info.encoding != ST_ENCODING_SIGN2)
{
st_warn("Do not support %s with 16-bit data. Forcing to Signed.",st_encodings_str[ft->info.encoding]);
ft->info.encoding = ST_ENCODING_SIGN2;
}
break;
case ST_SIZE_DWORD:
wBitsPerSample = 32;
if (ft->info.encoding != ST_ENCODING_SIGN2)
{
st_warn("Do not support %s with 16-bit data. Forcing to Signed.",st_encodings_str[ft->info.encoding]);
ft->info.encoding = ST_ENCODING_SIGN2;
}
break;
default:
st_warn("Do not support %s in WAV files. Forcing to Signed Words.",st_sizes_str[ft->info.size]);
ft->info.encoding = ST_ENCODING_SIGN2;
ft->info.size = ST_SIZE_WORD;
wBitsPerSample = 16;
break;
}
if (ft->info.encoding != ST_ENCODING_ADPCM &&
ft->info.encoding != ST_ENCODING_IMA_ADPCM &&
ft->info.encoding != ST_ENCODING_GSM)
{
rc = st_rawstartwrite(ft);
if (rc)
return rc;
}
wSamplesPerBlock = 1; /* common default for PCM data */
switch (ft->info.encoding)
{
case ST_ENCODING_UNSIGNED:
case ST_ENCODING_SIGN2:
wFormatTag = WAVE_FORMAT_PCM;
bytespersample = (wBitsPerSample + 7)/8;
wBlockAlign = wChannels * bytespersample;
break;
case ST_ENCODING_ALAW:
wFormatTag = WAVE_FORMAT_ALAW;
wBlockAlign = wChannels;
break;
case ST_ENCODING_ULAW:
wFormatTag = WAVE_FORMAT_MULAW;
wBlockAlign = wChannels;
break;
case ST_ENCODING_IMA_ADPCM:
if (wChannels>16)
{
st_fail("Channels(%d) must be <= 16\n",wChannels);
return ST_EOF;
}
wFormatTag = WAVE_FORMAT_IMA_ADPCM;
wBlockAlign = wChannels * 256; /* reasonable default */
wBitsPerSample = 4;
wExtSize = 2;
wSamplesPerBlock = ImaSamplesIn(0, wChannels, wBlockAlign, 0);
break;
case ST_ENCODING_ADPCM:
if (wChannels>16)
{
st_fail("Channels(%d) must be <= 16\n",wChannels);
return ST_EOF;
}
wFormatTag = WAVE_FORMAT_ADPCM;
wBlockAlign = wChannels * 128; /* reasonable default */
wBitsPerSample = 4;
wExtSize = 4+4*7; /* Ext fmt data length */
wSamplesPerBlock = AdpcmSamplesIn(0, wChannels, wBlockAlign, 0);
break;
case ST_ENCODING_GSM:
#ifdef HAVE_LIBGSM
if (wChannels!=1)
{
st_fail("Channels(%d) must be == 1\n",wChannels);
return ST_EOF;
}
wFormatTag = WAVE_FORMAT_GSM610;
/* wAvgBytesPerSec = 1625*(wSamplesPerSecond/8000.)+0.5; */
wBlockAlign=65;
wBitsPerSample=0; /* not representable as int */
wExtSize=2; /* length of format extension */
wSamplesPerBlock = 320;
#else
st_fail("sorry, no GSM6.10 support, recompile sox with gsm library");
return ST_EOF;
#endif
break;
}
wav->formatTag = wFormatTag;
wav->blockAlign = wBlockAlign;
wav->samplesPerBlock = wSamplesPerBlock;
if (!second_header) { /* adjust for blockAlign */
blocksWritten = wDataLength/wBlockAlign;
wDataLength = blocksWritten * wBlockAlign;
wSamplesWritten = blocksWritten * wSamplesPerBlock;
} else { /* fixup with real length */
wSamplesWritten = wav->numSamples;
switch(wFormatTag)
{
case WAVE_FORMAT_ADPCM:
case WAVE_FORMAT_IMA_ADPCM:
wDataLength = wav->dataLength;
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
/* intentional case fallthrough! */
#endif
default:
wSamplesWritten /= wChannels; /* because how rawwrite()'s work */
blocksWritten = (wSamplesWritten+wSamplesPerBlock-1)/wSamplesPerBlock;
wDataLength = blocksWritten * wBlockAlign;
}
}
#ifdef HAVE_LIBGSM
if (wFormatTag == WAVE_FORMAT_GSM610)
wDataLength = (wDataLength+1) & ~1; /*round up to even */
#endif
if (wFormatTag != WAVE_FORMAT_PCM)
wFmtSize += 2+wExtSize; /* plus ExtData */
wRiffLength = 4 + (8+wFmtSize) + (8+wDataLength);
if (wFormatTag != WAVE_FORMAT_PCM) /* PCM omits the "fact" chunk */
wRiffLength += (8+wFactSize);
/* wAvgBytesPerSec <-- this is BEFORE compression, isn't it? guess not. */
wAvgBytesPerSec = (double)wBlockAlign*ft->info.rate / (double)wSamplesPerBlock + 0.5;
/* figured out header info, so write it */
st_writes(ft, "RIFF");
st_writedw(ft, wRiffLength);
st_writes(ft, "WAVE");
st_writes(ft, "fmt ");
st_writedw(ft, wFmtSize);
st_writew(ft, wFormatTag);
st_writew(ft, wChannels);
st_writedw(ft, wSamplesPerSecond);
st_writedw(ft, wAvgBytesPerSec);
st_writew(ft, wBlockAlign);
st_writew(ft, wBitsPerSample); /* end info common to all fmts */
/* if not PCM, we need to write out wExtSize even if wExtSize=0 */
if (wFormatTag != WAVE_FORMAT_PCM)
st_writew(ft,wExtSize);
switch (wFormatTag)
{
int i;
case WAVE_FORMAT_IMA_ADPCM:
st_writew(ft, wSamplesPerBlock);
break;
case WAVE_FORMAT_ADPCM:
st_writew(ft, wSamplesPerBlock);
st_writew(ft, 7); /* nCoefs */
for (i=0; i<7; i++) {
st_writew(ft, iCoef[i][0]);
st_writew(ft, iCoef[i][1]);
}
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
st_writew(ft, wSamplesPerBlock);
break;
#endif
default:
break;
}
/* if not PCM, write the 'fact' chunk */
if (wFormatTag != WAVE_FORMAT_PCM){
st_writes(ft, "fact");
st_writedw(ft,wFactSize);
st_writedw(ft,wSamplesWritten);
}
st_writes(ft, "data");
st_writedw(ft, wDataLength); /* data chunk size */
if (!second_header) {
st_report("Writing Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wFormatTag), wChannels,
wChannels == 1 ? "" : "s", wSamplesPerSecond);
st_report(" %d byte/sec, %d block align, %d bits/samp",
wAvgBytesPerSec, wBlockAlign, wBitsPerSample);
} else {
st_report("Finished writing Wave file, %u data bytes %u samples\n",
wDataLength,wav->numSamples);
#ifdef HAVE_LIBGSM
if (wFormatTag == WAVE_FORMAT_GSM610){
st_report("GSM6.10 format: %u blocks %u padded samples %u padded data bytes\n",
blocksWritten, wSamplesWritten, wDataLength);
if (wav->gsmbytecount != wDataLength)
st_warn("help ! internal inconsistency - data_written %u gsmbytecount %u",
wDataLength, wav->gsmbytecount);
}
#endif
}
return ST_SUCCESS;
}
LONG st_wavwrite(ft, buf, len)
ft_t ft;
LONG *buf, len;
{
wav_t wav = (wav_t) ft->priv;
LONG save_len = len;
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 save_len - len;
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
wav->numSamples += len;
return wavgsmwrite(ft, buf, len);
break;
#endif
default:
wav->numSamples += len; /* must later be divided by wChannels */
return st_rawwrite(ft, buf, len);
}
}
int st_wavstopwrite(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
int rc = ST_SUCCESS;
/* Call this to flush out any remaining data. */
switch (wav->formatTag)
{
case WAVE_FORMAT_IMA_ADPCM:
case WAVE_FORMAT_ADPCM:
rc = xxxAdpcmWriteBlock(ft);
break;
#ifdef HAVE_LIBGSM
case WAVE_FORMAT_GSM610:
wavgsmstopwrite(ft);
break;
#endif
default:
rc = st_rawstopwrite(ft);
}
if (wav->packet) free(wav->packet);
if (wav->samples) free(wav->samples);
if (wav->iCoefs) free(wav->iCoefs);
/* Now that we've free()'d memory, return with errors if needed */
if (rc)
return rc;
/* 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->seekable)
return ST_EOF;
if (fseek(ft->fp, 0L, SEEK_SET) != 0)
{
st_fail("Sorry, can't rewind output file to rewrite .wav header.");
return ST_EOF;
}
wavwritehdr(ft, 1);
return (ST_SUCCESS);
}
/*
* Return a string corresponding to the wave format type.
*/
static char *
wav_format_str(wFormatTag)
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";
}
}