ref: 3d18ea29658bfb1ac4be84d4ae0f4660199ca76d
dir: /src/wav.c/
/*
* Microsoft's WAVE sound format driver
*
* Copyright 1998-2006 Chris Bagwell and SoX Contributors
* Copyright 1991 Lance Norskog And Sundry Contributors
* Copyright 1992 Rick Richardson
* Copyright 1997 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>
#include <stdlib.h>
#include <stdio.h>
#ifdef HAVE_UNISTD_H
#include <unistd.h> /* For SEEK_* defines if not found in stdio */
#endif
#include "st_i.h"
#include "wav.h"
#include "ima_rw.h"
#include "adpcm.h"
#ifdef EXTERNAL_GSM
#include <gsm/gsm.h>
#else
#include "libgsm/gsm.h"
#endif
/* To allow padding to samplesPerBlock. Works, but currently never true. */
static st_size_t pad_nsamps = false;
/* Private data for .wav file */
typedef struct wavstuff {
st_size_t numSamples; /* samples/channel reading: starts at total count and decremented */
/* writing: starts at 0 and counts samples written */
st_size_t dataLength; /* needed for ADPCM writing */
unsigned short formatTag; /* What type of encoding file is using */
unsigned short samplesPerBlock;
unsigned short blockAlign;
st_size_t dataStart; /* need to for seeking */
int found_cooledit;
/* 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 */
gsm gsmhandle;
gsm_signal *gsmsample;
int gsmindex;
st_size_t gsmbytecount; /* counts bytes written to data block */
} *wav_t;
static char *wav_format_str(unsigned wFormatTag);
static int wavwritehdr(ft_t, int);
/****************************************************************************/
/* IMA ADPCM Support Functions Section */
/****************************************************************************/
/*
*
* ImaAdpcmReadBlock - Grab and decode complete block of samples
*
*/
static unsigned short ImaAdpcmReadBlock(ft_t ft)
{
wav_t wav = (wav_t) ft->priv;
int bytesRead;
int samplesThisBlock;
/* Pull in the packet and check the header */
bytesRead = st_readbuf(ft, wav->packet, 1, 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 = ImaSamplesIn(0, ft->signal.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->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(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 = st_readbuf(ft, wav->packet, 1, 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 = AdpcmSamplesIn(0, ft->signal.channels, bytesRead, 0);
if (samplesThisBlock == 0)
{
st_warn("Premature EOF on .wav input file");
return 0;
}
}
errmsg = AdpcmBlockExpandI(ft->signal.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_t ft)
{
wav_t wav = (wav_t) ft->priv;
int 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) {
AdpcmBlockMashI(chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, wav->blockAlign);
}else{ /* WAVE_FORMAT_IMA_ADPCM */
ImaBlockMashI(chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, 9);
}
/* write the compressed packet */
if (st_writebuf(ft, wav->packet, wav->blockAlign, 1) != 1)
{
st_fail_errno(ft,ST_EOF,"write error");
return (ST_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 (ST_SUCCESS);
}
/****************************************************************************/
/* WAV GSM6.10 support functions */
/****************************************************************************/
/* create the gsm object, malloc buffer for 160*2 samples */
static int wavgsminit(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_errno(ft,ST_EOF,"cannot create GSM object");
return (ST_EOF);
}
if(gsm_option(wav->gsmhandle,GSM_OPT_WAV49,&valueP) == -1){
st_fail_errno(ft,ST_EOF,"error setting gsm_option for WAV49 format. Recompile gsm library with -DWAV49 option and relink sox");
return (ST_EOF);
}
wav->gsmsample=(gsm_signal*)xmalloc(sizeof(gsm_signal)*160*2);
wav->gsmindex=0;
return (ST_SUCCESS);
}
/*destroy the gsm object and free the buffer */
static void wavgsmdestroy(ft_t ft)
{
wav_t wav = (wav_t) ft->priv;
gsm_destroy(wav->gsmhandle);
free(wav->gsmsample);
}
static st_size_t wavgsmread(ft_t ft, st_sample_t *buf, st_size_t len)
{
wav_t wav = (wav_t) ft->priv;
size_t done=0;
int bytes;
gsm_byte frame[65];
ft->st_errno = ST_SUCCESS;
/* copy out any samples left from the last call */
while(wav->gsmindex && (wav->gsmindex<160*2) && (done < len))
buf[done++]=ST_SIGNED_WORD_TO_SAMPLE(wav->gsmsample[wav->gsmindex++],);
/* read and decode loop, possibly leaving some samples in wav->gsmsample */
while (done < len) {
wav->gsmindex=0;
bytes = st_readbuf(ft, frame, 1, 65);
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_errno(ft,ST_EOF,"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_errno(ft,ST_EOF,"error during gsm decode");
return 0;
}
while ((wav->gsmindex <160*2) && (done < len)){
buf[done++]=ST_SIGNED_WORD_TO_SAMPLE(wav->gsmsample[(wav->gsmindex)++],);
}
}
return done;
}
static int wavgsmflush(ft_t ft)
{
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 (st_writebuf(ft, frame, 1, 65) != 65)
{
st_fail_errno(ft,ST_EOF,"write error");
return (ST_EOF);
}
wav->gsmbytecount += 65;
wav->gsmindex = 0;
return (ST_SUCCESS);
}
static st_size_t wavgsmwrite(ft_t ft, const st_sample_t *buf, st_size_t len)
{
wav_t wav = (wav_t) ft->priv;
size_t done = 0;
int rc;
ft->st_errno = ST_SUCCESS;
while (done < len) {
while ((wav->gsmindex < 160*2) && (done < len))
wav->gsmsample[(wav->gsmindex)++] =
ST_SAMPLE_TO_SIGNED_WORD(buf[done++], ft->clippedCount);
if (wav->gsmindex < 160*2)
break;
rc = wavgsmflush(ft);
if (rc)
return 0;
}
return done;
}
static void wavgsmstopwrite(ft_t ft)
{
wav_t wav = (wav_t) ft->priv;
ft->st_errno = ST_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(st_writeb(ft, 0))
st_fail_errno(ft,ST_EOF,"write error");
else
wav->gsmbytecount += 1;
}
wavgsmdestroy(ft);
}
/****************************************************************************/
/* General Sox WAV file code */
/****************************************************************************/
static int findChunk(ft_t ft, const char *Label, st_size_t *len)
{
char magic[5];
for (;;)
{
if (st_reads(ft, magic, 4) == ST_EOF)
{
st_fail_errno(ft, ST_EHDR, "WAVE file has missing %s chunk",
Label);
return ST_EOF;
}
st_debug("WAV Chunk %s", magic);
if (st_readdw(ft, len) == ST_EOF)
{
st_fail_errno(ft, ST_EHDR, "WAVE file %s chunk is too short",
magic);
return ST_EOF;
}
if (strncmp(Label, magic, 4) == 0)
break; /* Found the data chunk */
/* skip to next chunk */
if (st_seeki(ft, *len, SEEK_CUR) != ST_SUCCESS)
{
st_fail_errno(ft,ST_EHDR,
"WAV chunk appears to have invalid size %d.", *len);
return ST_EOF;
}
}
return ST_SUCCESS;
}
/*
* 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 st_wavstartread(ft_t ft)
{
wav_t wav = (wav_t) ft->priv;
char magic[5];
uint32_t len;
int rc;
/* 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 */
st_size_t bytesPerBlock = 0;
int bytespersample; /* bytes per sample (per channel */
char text[256];
uint32_t dwLoopPos;
ft->st_errno = ST_SUCCESS;
if (st_reads(ft, magic, 4) == ST_EOF || (strncmp("RIFF", magic, 4) != 0 &&
strncmp("RIFX", magic, 4) != 0))
{
st_fail_errno(ft,ST_EHDR,"WAVE: RIFF header not found");
return ST_EOF;
}
/* RIFX is a Big-endian RIFF */
if (strncmp("RIFX", magic, 4) == 0)
{
st_debug("Found RIFX header, swapping bytes");
ft->signal.swap_bytes = ST_IS_LITTLEENDIAN;
}
st_readdw(ft, &dwRiffLength);
if (st_reads(ft, magic, 4) == ST_EOF || strncmp("WAVE", magic, 4))
{
st_fail_errno(ft,ST_EHDR,"WAVE header not found");
return ST_EOF;
}
/* Now look for the format chunk */
if (findChunk(ft, "fmt ", &len) == ST_EOF)
{
st_fail_errno(ft,ST_EHDR,"WAVE chunk fmt not found");
return ST_EOF;
}
wFmtSize = len;
if (wFmtSize < 16)
{
st_fail_errno(ft,ST_EHDR,"WAVE file fmt chunk is too short");
return ST_EOF;
}
st_readw(ft, &(wav->formatTag));
st_readw(ft, &wChannels);
st_readdw(ft, &dwSamplesPerSecond);
st_readdw(ft, &dwAvgBytesPerSec); /* Average bytes/second */
st_readw(ft, &(wav->blockAlign)); /* Block align */
st_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)
{
st_fail_errno(ft,ST_EHDR,"WAVE file fmt chunk is too short");
return ST_EOF;
}
st_readw(ft, &extensionSize);
len -= 2;
if (extensionSize < 22)
{
st_fail_errno(ft,ST_EHDR,"WAVE file fmt chunk is too short");
return ST_EOF;
}
st_readw(ft, &numberOfValidBits);
st_readdw(ft, &speakerPositionMask);
st_readw(ft, &subFormatTag);
for (i = 0; i < 14; ++i) st_readb(ft, &dummyByte);
len -= 22;
if (numberOfValidBits != wBitsPerSample)
{
st_fail_errno(ft,ST_EHDR,"WAVE file fmt with padded samples is not supported yet");
return ST_EOF;
}
wav->formatTag = subFormatTag;
}
switch (wav->formatTag)
{
case WAVE_FORMAT_UNKNOWN:
st_fail_errno(ft,ST_EHDR,"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->signal.encoding != ST_ENCODING_UNKNOWN && ft->signal.encoding != ST_ENCODING_UNSIGNED &&
ft->signal.encoding != ST_ENCODING_SIGN2)
st_report("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->signal.encoding == ST_ENCODING_UNKNOWN || ft->signal.encoding == ST_ENCODING_IMA_ADPCM)
ft->signal.encoding = ST_ENCODING_IMA_ADPCM;
else
st_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_ADPCM:
if (ft->signal.encoding == ST_ENCODING_UNKNOWN || ft->signal.encoding == ST_ENCODING_ADPCM)
ft->signal.encoding = ST_ENCODING_ADPCM;
else
st_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_IEEE_FLOAT:
if (ft->signal.encoding == ST_ENCODING_UNKNOWN || ft->signal.encoding == ST_ENCODING_FLOAT)
ft->signal.encoding = ST_ENCODING_FLOAT;
else
st_report("User options overriding encoding read in .wav header");
/* Needed by rawread() functions */
rc = st_rawstartread(ft);
if (rc)
return rc;
break;
case WAVE_FORMAT_ALAW:
if (ft->signal.encoding == ST_ENCODING_UNKNOWN || ft->signal.encoding == ST_ENCODING_ALAW)
ft->signal.encoding = ST_ENCODING_ALAW;
else
st_report("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->signal.encoding == ST_ENCODING_UNKNOWN || ft->signal.encoding == ST_ENCODING_ULAW)
ft->signal.encoding = ST_ENCODING_ULAW;
else
st_report("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_errno(ft,ST_EHDR,"Sorry, this WAV file is in OKI ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_DIGISTD:
st_fail_errno(ft,ST_EHDR,"Sorry, this WAV file is in Digistd format.");
return ST_EOF;
case WAVE_FORMAT_DIGIFIX:
st_fail_errno(ft,ST_EHDR,"Sorry, this WAV file is in Digifix format.");
return ST_EOF;
case WAVE_FORMAT_DOLBY_AC2:
st_fail_errno(ft,ST_EHDR,"Sorry, this WAV file is in Dolby AC2 format.");
return ST_EOF;
case WAVE_FORMAT_GSM610:
if (ft->signal.encoding == ST_ENCODING_UNKNOWN || ft->signal.encoding == ST_ENCODING_GSM )
ft->signal.encoding = ST_ENCODING_GSM;
else
st_report("User options overriding encoding read in .wav header");
break;
case WAVE_FORMAT_ROCKWELL_ADPCM:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in Rockwell ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_ROCKWELL_DIGITALK:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in Rockwell DIGITALK format.");
return ST_EOF;
case WAVE_FORMAT_G721_ADPCM:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in G.721 ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_G728_CELP:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in G.728 CELP format.");
return ST_EOF;
case WAVE_FORMAT_MPEG:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in MPEG format.");
return ST_EOF;
case WAVE_FORMAT_MPEGLAYER3:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in MPEG Layer 3 format.");
return ST_EOF;
case WAVE_FORMAT_G726_ADPCM:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in G.726 ADPCM format.");
return ST_EOF;
case WAVE_FORMAT_G722_ADPCM:
st_fail_errno(ft,ST_EOF,"Sorry, this WAV file is in G.722 ADPCM format.");
return ST_EOF;
default: st_fail_errno(ft,ST_EOF,"WAV file has unknown format type of %x",wav->formatTag);
return ST_EOF;
}
/* User options take precedence */
if (ft->signal.channels == 0 || ft->signal.channels == wChannels)
ft->signal.channels = wChannels;
else
st_report("User options overriding channels read in .wav header");
if (ft->signal.rate == 0 || ft->signal.rate == dwSamplesPerSecond)
ft->signal.rate = dwSamplesPerSecond;
else
st_report("User options overriding rate read in .wav header");
wav->iCoefs = 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) {
st_readw(ft, &wExtSize);
len -= 2;
} else {
st_warn("wave header missing FmtExt chunk");
}
}
if (wExtSize > len)
{
st_fail_errno(ft,ST_EOF,"wave header error: wExtSize inconsistent with wFmtLen");
return ST_EOF;
}
switch (wav->formatTag)
{
case WAVE_FORMAT_ADPCM:
if (wExtSize < 4)
{
st_fail_errno(ft,ST_EOF,"format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 4);
return ST_EOF;
}
if (wBitsPerSample != 4)
{
st_fail_errno(ft,ST_EOF,"Can only handle 4-bit MS ADPCM in wav files");
return ST_EOF;
}
st_readw(ft, &(wav->samplesPerBlock));
bytesPerBlock = AdpcmBytesPerBlock(ft->signal.channels, wav->samplesPerBlock);
if (bytesPerBlock > wav->blockAlign)
{
st_fail_errno(ft,ST_EOF,"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_errno(ft,ST_EOF,"ADPCM file nCoefs (%.4hx) makes no sense", wav->nCoefs);
return ST_EOF;
}
wav->packet = (unsigned char *)xmalloc(wav->blockAlign);
len -= 4;
if (wExtSize < 4 + 4*wav->nCoefs)
{
st_fail_errno(ft,ST_EOF,"wave header error: wExtSize(%d) too small for nCoefs(%d)", wExtSize, wav->nCoefs);
return ST_EOF;
}
wav->samples = (short *)xmalloc(wChannels*wav->samplesPerBlock*sizeof(short));
/* nCoefs, iCoefs used by adpcm.c */
wav->iCoefs = (short *)xmalloc(wav->nCoefs * 2 * sizeof(short));
{
int i, errct=0;
for (i=0; len>=2 && i < 2*wav->nCoefs; i++) {
st_readw(ft, (unsigned short *)&(wav->iCoefs[i]));
len -= 2;
if (i<14) errct += (wav->iCoefs[i] != iCoef[i/2][i%2]);
/* st_debug("iCoefs[%2d] %4d",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_errno(ft,ST_EOF,"format[%s]: expects wExtSize >= %d",
wav_format_str(wav->formatTag), 2);
return ST_EOF;
}
if (wBitsPerSample != 4)
{
st_fail_errno(ft,ST_EOF,"Can only handle 4-bit IMA ADPCM in wav files");
return ST_EOF;
}
st_readw(ft, &(wav->samplesPerBlock));
bytesPerBlock = ImaBytesPerBlock(ft->signal.channels, wav->samplesPerBlock);
if (bytesPerBlock > wav->blockAlign || wav->samplesPerBlock%8 != 1)
{
st_fail_errno(ft,ST_EOF,"format[%s]: samplesPerBlock(%d) incompatible with blockAlign(%d)",
wav_format_str(wav->formatTag), wav->samplesPerBlock, wav->blockAlign);
return ST_EOF;
}
wav->packet = (unsigned char *)xmalloc(wav->blockAlign);
len -= 2;
wav->samples = (short *)xmalloc(wChannels*wav->samplesPerBlock*sizeof(short));
bytespersample = ST_SIZE_WORD; /* AFTER de-compression */
break;
/* GSM formats have extended fmt chunk. Check for those cases. */
case WAVE_FORMAT_GSM610:
if (wExtSize < 2)
{
st_fail_errno(ft,ST_EOF,"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_errno(ft,ST_EOF,"format[%s]: expects blockAlign(%d) = %d",
wav_format_str(wav->formatTag), wav->blockAlign, 65);
return ST_EOF;
}
if (wav->samplesPerBlock != 320)
{
st_fail_errno(ft,ST_EOF,"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;
default:
bytespersample = (wBitsPerSample + 7)/8;
}
switch (bytespersample)
{
case ST_SIZE_BYTE:
/* User options take precedence */
if (ft->signal.size == -1 || ft->signal.size == ST_SIZE_BYTE)
ft->signal.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->signal.encoding == ST_ENCODING_UNKNOWN)
ft->signal.encoding = ST_ENCODING_UNSIGNED;
break;
case ST_SIZE_WORD:
if (ft->signal.size == -1 || ft->signal.size == ST_SIZE_WORD)
ft->signal.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->signal.encoding == ST_ENCODING_UNKNOWN)
ft->signal.encoding = ST_ENCODING_SIGN2;
break;
case ST_SIZE_24BIT:
if (ft->signal.size == -1 || ft->signal.size == ST_SIZE_24BIT)
ft->signal.size = ST_SIZE_24BIT;
else
st_warn("User options overriding size read in .wav header");
/* Now we have enough information to set default encodings. */
if (ft->signal.encoding == ST_ENCODING_UNKNOWN)
ft->signal.encoding = ST_ENCODING_SIGN2;
break;
case ST_SIZE_DWORD:
if (ft->signal.size == -1 || ft->signal.size == ST_SIZE_DWORD)
ft->signal.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->signal.encoding == ST_ENCODING_UNKNOWN)
ft->signal.encoding = ST_ENCODING_SIGN2;
break;
default:
st_fail_errno(ft,ST_EOF,"Sorry, don't understand .wav size");
return ST_EOF;
}
/* Skip anything left over from fmt chunk */
st_seeki(ft, 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) == ST_EOF)
{
st_fail_errno(ft, ST_EOF, "Could not find data chunk.");
return ST_EOF;
}
dwDataLength = len;
/* Data starts here */
wav->dataStart = st_tell(ft);
switch (wav->formatTag)
{
case WAVE_FORMAT_ADPCM:
wav->numSamples =
AdpcmSamplesIn(dwDataLength, ft->signal.channels,
wav->blockAlign, wav->samplesPerBlock);
/*st_debug("datalen %d, numSamples %d",dwDataLength, wav->numSamples);*/
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
ft->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 =
ImaSamplesIn(dwDataLength, ft->signal.channels,
wav->blockAlign, wav->samplesPerBlock);
/*st_debug("datalen %d, numSamples %d",dwDataLength, wav->numSamples);*/
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
initImaTable();
ft->length = wav->numSamples*ft->signal.channels;
break;
case WAVE_FORMAT_GSM610:
wav->numSamples = ((dwDataLength / wav->blockAlign) * wav->samplesPerBlock);
wavgsminit(ft);
ft->length = wav->numSamples*ft->signal.channels;
break;
default:
wav->numSamples = dwDataLength/ft->signal.size/ft->signal.channels;
ft->length = wav->numSamples*ft->signal.channels;
}
st_debug("Reading Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wav->formatTag), ft->signal.channels,
wChannels == 1 ? "" : "s", dwSamplesPerSecond);
st_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:
st_debug(" %d Extsize, %d Samps/block, %d bytes/block %d Num Coefs, %d Samps/chan",
wExtSize,wav->samplesPerBlock,bytesPerBlock,wav->nCoefs,
wav->numSamples);
break;
case WAVE_FORMAT_IMA_ADPCM:
st_debug(" %d Extsize, %d Samps/block, %d bytes/block %d Samps/chan",
wExtSize, wav->samplesPerBlock, bytesPerBlock,
wav->numSamples);
break;
case WAVE_FORMAT_GSM610:
st_debug("GSM .wav: %d Extsize, %d Samps/block, %d Samples/chan",
wExtSize, wav->samplesPerBlock, wav->numSamples);
break;
default:
st_debug(" %d Samps/chans", wav->numSamples);
}
/* Horrible way to find Cool Edit marker points. Taken from Quake source*/
ft->loops[0].start = -1;
wav->found_cooledit = 0;
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) & ~1;
if (st_seeki(ft, len, SEEK_CUR) == ST_SUCCESS &&
findChunk(ft, "LIST", &len) != ST_EOF)
{
wav->found_cooledit = 1;
ft->comment = (char*)xmalloc(256);
/* Initialize comment to a NULL string */
ft->comment[0] = 0;
while(!st_eof(ft))
{
if (st_reads(ft,magic,4) == ST_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", 4) == 0)
{
/*Skip*/
st_debug("Type INFO");
}
else if (strncmp(magic, "adtl", 4) == 0)
{
/* Skip */
st_debug("Type adtl");
}
else
{
if (st_readdw(ft,&len) == ST_EOF)
break;
if (strncmp(magic,"ICRD",4) == 0)
{
st_debug("Chunk ICRD");
if (len > 254)
{
st_warn("Possible buffer overflow hack attack (ICRD)!");
break;
}
st_reads(ft,text,len);
if (strlen(ft->comment) + strlen(text) < 254)
{
if (ft->comment[0] != 0)
strcat(ft->comment,"\n");
strcat(ft->comment,text);
}
if (strlen(text) < len)
st_seeki(ft, len - strlen(text), SEEK_CUR);
}
else if (strncmp(magic,"ISFT",4) == 0)
{
st_debug("Chunk ISFT");
if (len > 254)
{
st_warn("Possible buffer overflow hack attack (ISFT)!");
break;
}
st_reads(ft,text,len);
if (strlen(ft->comment) + strlen(text) < 254)
{
if (ft->comment[0] != 0)
strcat(ft->comment,"\n");
strcat(ft->comment,text);
}
if (strlen(text) < len)
st_seeki(ft, len - strlen(text), SEEK_CUR);
}
else if (strncmp(magic,"cue ",4) == 0)
{
st_debug("Chunk cue ");
st_seeki(ft,len-4,SEEK_CUR);
st_readdw(ft,&dwLoopPos);
ft->loops[0].start = dwLoopPos;
}
else if (strncmp(magic,"ltxt",4) == 0)
{
st_debug("Chunk ltxt");
st_readdw(ft,&dwLoopPos);
ft->loops[0].length = dwLoopPos - ft->loops[0].start;
if (len > 4)
st_seeki(ft, len - 4, SEEK_CUR);
}
else
{
st_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) & ~1;
st_seeki(ft, len, SEEK_CUR);
}
}
}
}
st_clearerr(ft);
st_seeki(ft,wav->dataStart,SEEK_SET);
}
return ST_SUCCESS;
}
/*
* Read up to len samples from file.
* Convert to signed longs.
* Place in buf[].
* Return number of samples read.
*/
static st_size_t st_wavread(ft_t ft, st_sample_t *buf, st_size_t len)
{
wav_t wav = (wav_t) ft->priv;
st_size_t done;
ft->st_errno = ST_SUCCESS;
/* If file is in ADPCM encoding then read in multiple blocks else */
/* read as much as possible and return quickly. */
switch (ft->signal.encoding)
{
case ST_ENCODING_IMA_ADPCM:
case ST_ENCODING_ADPCM:
/* See reason for cooledit check in comments below */
if (wav->found_cooledit && 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 st_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++ = ST_SIGNED_WORD_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 ST_ENCODING_GSM:
/* See reason for cooledit check in comments below */
if (wav->found_cooledit && len > wav->numSamples*ft->signal.channels)
len = (wav->numSamples*ft->signal.channels);
done = wavgsmread(ft, buf, len);
if (done == 0 && wav->numSamples != 0)
st_warn("Premature EOF on .wav input file");
break;
default: /* assume PCM or float encoding */
/* Cooledit seems to put a non-standard IFF LIST at
* the end of the file. When this is detected,
* go ahead and only read in the reported size
* of data chunk so the LIST data is not treated
* as audio.
* In other cases, go ahead and read unit EOF
* This allows us to process WAV files that are
* greater then 2Gig and can't be represented
* by the 32-bit size field.
*/
if (wav->found_cooledit && len > wav->numSamples*ft->signal.channels)
len = (wav->numSamples*ft->signal.channels);
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");
}
/* 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 st_wavstopread(ft_t ft)
{
wav_t wav = (wav_t) ft->priv;
int rc = ST_SUCCESS;
ft->st_errno = ST_SUCCESS;
free(wav->packet);
free(wav->samples);
free(wav->iCoefs);
free(ft->comment);
ft->comment = NULL;
switch (ft->signal.encoding)
{
case ST_ENCODING_GSM:
wavgsmdestroy(ft);
break;
case ST_ENCODING_IMA_ADPCM:
case ST_ENCODING_ADPCM:
break;
default:
/* Needed for rawread() */
rc = st_rawstopread(ft);
}
return rc;
}
static int st_wavstartwrite(ft_t ft)
{
wav_t wav = (wav_t) ft->priv;
int rc;
ft->st_errno = ST_SUCCESS;
if (ft->signal.encoding != ST_ENCODING_ADPCM &&
ft->signal.encoding != ST_ENCODING_IMA_ADPCM &&
ft->signal.encoding != ST_ENCODING_GSM)
{
rc = st_rawstartwrite(ft);
if (rc)
return rc;
}
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)
{
size_t 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->signal.channels; ch++)
wav->state[ch] = 0;
sbsize = ft->signal.channels * wav->samplesPerBlock;
wav->packet = (unsigned char *)xmalloc(wav->blockAlign);
wav->samples = (short *)xmalloc(sbsize*sizeof(short));
wav->sampleTop = wav->samples + sbsize;
wav->samplePtr = wav->samples;
break;
case WAVE_FORMAT_GSM610:
wavgsminit(ft);
break;
default:
break;
}
return ST_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
*/
static int wavwritehdr(ft_t ft, int second_header)
{
wav_t wav = (wav_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=0x7ffff000L; /* 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;
bool isExtensible = false; /* WAVE_FORMAT_EXTENSIBLE? */
dwSamplesPerSecond = ft->signal.rate;
wChannels = ft->signal.channels;
/* Check to see if encoding is ADPCM or not. If ADPCM
* possibly override the size to be bytes. It isn't needed
* by this routine will look nicer (and more correct)
* on verbose output.
*/
if ((ft->signal.encoding == ST_ENCODING_ADPCM ||
ft->signal.encoding == ST_ENCODING_IMA_ADPCM ||
ft->signal.encoding == ST_ENCODING_GSM) &&
ft->signal.size != ST_SIZE_BYTE)
{
st_report("Overriding output size to bytes for compressed data.");
ft->signal.size = ST_SIZE_BYTE;
}
switch (ft->signal.size)
{
case ST_SIZE_BYTE:
wBitsPerSample = 8;
if (ft->signal.encoding != ST_ENCODING_UNSIGNED &&
ft->signal.encoding != ST_ENCODING_ULAW &&
ft->signal.encoding != ST_ENCODING_ALAW &&
ft->signal.encoding != ST_ENCODING_GSM &&
ft->signal.encoding != ST_ENCODING_ADPCM &&
ft->signal.encoding != ST_ENCODING_IMA_ADPCM)
{
st_report("Do not support %s with 8-bit data. Forcing to unsigned",st_encodings_str[(unsigned char)ft->signal.encoding]);
ft->signal.encoding = ST_ENCODING_UNSIGNED;
}
break;
case ST_SIZE_WORD:
wBitsPerSample = 16;
if (ft->signal.encoding != ST_ENCODING_SIGN2)
{
st_report("Do not support %s with 16-bit data. Forcing to Signed.",st_encodings_str[(unsigned char)ft->signal.encoding]);
ft->signal.encoding = ST_ENCODING_SIGN2;
}
break;
case ST_SIZE_24BIT:
wBitsPerSample = 24;
if (ft->signal.encoding != ST_ENCODING_SIGN2)
{
st_report("Do not support %s with 24-bit data. Forcing to Signed.",st_encodings_str[(unsigned char)ft->signal.encoding]);
ft->signal.encoding = ST_ENCODING_SIGN2;
}
break;
case ST_SIZE_DWORD:
wBitsPerSample = 32;
if (ft->signal.encoding != ST_ENCODING_SIGN2 &&
ft->signal.encoding != ST_ENCODING_FLOAT)
{
st_report("Do not support %s with 32-bit data. Forcing to Signed.",st_encodings_str[(unsigned char)ft->signal.encoding]);
ft->signal.encoding = ST_ENCODING_SIGN2;
}
break;
default:
st_report("Do not support %s in WAV files. Forcing to Signed Words.",st_sizes_str[(unsigned char)ft->signal.size]);
ft->signal.encoding = ST_ENCODING_SIGN2;
ft->signal.size = ST_SIZE_WORD;
wBitsPerSample = 16;
break;
}
wSamplesPerBlock = 1; /* common default for PCM data */
switch (ft->signal.encoding)
{
case ST_ENCODING_UNSIGNED:
case ST_ENCODING_SIGN2:
wFormatTag = WAVE_FORMAT_PCM;
bytespersample = (wBitsPerSample + 7)/8;
wBlockAlign = wChannels * bytespersample;
break;
case ST_ENCODING_FLOAT:
wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
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_errno(ft,ST_EOF,"Channels(%d) must be <= 16",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_errno(ft,ST_EOF,"Channels(%d) must be <= 16",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:
if (wChannels!=1)
{
st_report("Overriding GSM audio from %d channel to 1",wChannels);
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) { /* adjust for blockAlign */
blocksWritten = dwDataLength/wBlockAlign;
dwDataLength = blocksWritten * wBlockAlign;
dwSamplesWritten = blocksWritten * wSamplesPerBlock;
} else { /* fixup with real length */
dwSamplesWritten = wav->numSamples;
switch(wFormatTag)
{
case WAVE_FORMAT_ADPCM:
case WAVE_FORMAT_IMA_ADPCM:
dwDataLength = wav->dataLength;
break;
case WAVE_FORMAT_GSM610:
/* intentional case fallthrough! */
default:
blocksWritten = (dwSamplesWritten+wSamplesPerBlock-1)/wSamplesPerBlock;
dwDataLength = blocksWritten * wBlockAlign;
}
}
if (wFormatTag == WAVE_FORMAT_GSM610)
dwDataLength = (dwDataLength+1) & ~1; /*round up to even */
if ((wFormatTag == WAVE_FORMAT_PCM && wBitsPerSample > 16) || wChannels > 2)
{
isExtensible = 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 then we think, assume they are
* asking to write a RIFX file.
*/
if (ft->signal.swap_bytes != ST_IS_BIGENDIAN)
{
if (!second_header)
st_report("Requested to swap bytes so writing RIFX header");
st_writes(ft, "RIFX");
}
else
st_writes(ft, "RIFF");
st_writedw(ft, wRiffLength);
st_writes(ft, "WAVE");
st_writes(ft, "fmt ");
st_writedw(ft, wFmtSize);
st_writew(ft, isExtensible? WAVE_FORMAT_EXTENSIBLE : wFormatTag);
st_writew(ft, wChannels);
st_writedw(ft, dwSamplesPerSecond);
st_writedw(ft, dwAvgBytesPerSec);
st_writew(ft, wBlockAlign);
st_writew(ft, wBitsPerSample); /* end info common to all fmts */
if (isExtensible)
{
size_t i;
static const char guid[14] = "\x00\x00\x00\x00\x10\x00\x80\x00\x00\xAA\x00\x38\x9B\x71";
st_writew(ft, 22);
st_writew(ft, wBitsPerSample); /* No padding in container */
st_writedw(ft, 0); /* Speaker mapping not specified */
st_writew(ft, wFormatTag);
for (i = 0; i < array_length(guid); ++i)
{
st_writeb(ft, guid[i]);
}
}
else
/* 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;
case WAVE_FORMAT_GSM610:
st_writew(ft, wSamplesPerBlock);
break;
default:
break;
}
/* if not PCM, write the 'fact' chunk */
if (isExtensible || wFormatTag != WAVE_FORMAT_PCM){
st_writes(ft, "fact");
st_writedw(ft,dwFactSize);
st_writedw(ft,dwSamplesWritten);
}
st_writes(ft, "data");
st_writedw(ft, dwDataLength); /* data chunk size */
if (!second_header) {
st_debug("Writing Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wFormatTag), wChannels,
wChannels == 1 ? "" : "s", dwSamplesPerSecond);
st_debug(" %d byte/sec, %d block align, %d bits/samp",
dwAvgBytesPerSec, wBlockAlign, wBitsPerSample);
} else {
st_debug("Finished writing Wave file, %u data bytes %u samples",
dwDataLength,wav->numSamples);
if (wFormatTag == WAVE_FORMAT_GSM610){
st_debug("GSM6.10 format: %u blocks %u padded samples %u padded data bytes",
blocksWritten, dwSamplesWritten, dwDataLength);
if (wav->gsmbytecount != dwDataLength)
st_warn("help ! internal inconsistency - data_written %u gsmbytecount %u",
dwDataLength, wav->gsmbytecount);
}
}
return ST_SUCCESS;
}
static st_size_t st_wavwrite(ft_t ft, const st_sample_t *buf, st_size_t len)
{
wav_t wav = (wav_t) ft->priv;
st_ssize_t total_len = len;
ft->st_errno = ST_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 = st_rawwrite(ft, buf, len);
wav->numSamples += (len/ft->signal.channels);
return len;
}
}
static int st_wavstopwrite(ft_t ft)
{
wav_t wav = (wav_t) ft->priv;
ft->st_errno = ST_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->iCoefs);
/* Flush any remaining data */
if (wav->formatTag != WAVE_FORMAT_IMA_ADPCM &&
wav->formatTag != WAVE_FORMAT_ADPCM &&
wav->formatTag != WAVE_FORMAT_GSM610)
st_rawstopwrite(ft);
/* 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 (st_seeki(ft, 0L, SEEK_SET) != 0)
{
st_fail_errno(ft,ST_EOF,"Can't rewind output file to rewrite .wav header.");
return ST_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 st_wavseek(ft_t ft, st_size_t offset)
{
wav_t wav = (wav_t) ft->priv;
int new_offset, channel_block, alignment;
switch (wav->formatTag)
{
case WAVE_FORMAT_IMA_ADPCM:
case WAVE_FORMAT_ADPCM:
st_fail_errno(ft,ST_ENOTSUP,"ADPCM not supported");
break;
case WAVE_FORMAT_GSM610:
{
st_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->st_errno = st_seeki(ft, gsmoff + wav->dataStart, SEEK_SET);
if (ft->st_errno != ST_SUCCESS)
return ST_EOF;
/* offset is in samples */
new_offset = offset;
alignment = offset % wav->samplesPerBlock;
if (alignment != 0)
new_offset += (wav->samplesPerBlock - alignment);
wav->numSamples = ft->length - (new_offset / ft->signal.channels);
}
break;
default:
new_offset = offset * ft->signal.size;
/* Make sure request aligns to a channel block (ie left+right) */
channel_block = ft->signal.channels * ft->signal.size;
alignment = new_offset % channel_block;
/* Most common mistaken is to compute something like
* "skip everthing upto and including this sample" so
* advance to next sample block in this case.
*/
if (alignment != 0)
new_offset += (channel_block - alignment);
new_offset += wav->dataStart;
ft->st_errno = st_seeki(ft, new_offset, SEEK_SET);
if( ft->st_errno == ST_SUCCESS )
wav->numSamples = (ft->length / ft->signal.channels) -
(new_offset / ft->signal.size / ft->signal.channels);
}
return(ft->st_errno);
}
/* Microsoft RIFF */
static const char *wavnames[] = {
"wav",
NULL
};
static st_format_t st_wav_format = {
wavnames,
NULL,
ST_FILE_STEREO | ST_FILE_SEEK | ST_FILE_LIT_END,
st_wavstartread,
st_wavread,
st_wavstopread,
st_wavstartwrite,
st_wavwrite,
st_wavstopwrite,
st_wavseek
};
const st_format_t *st_wav_format_fn()
{
return &st_wav_format;
}