ref: 0dc38173e5aec91f1c74cda9e43a201af6cb525a
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:
*
* 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. Info on these formats
* was taken from the xanim project, written by
* Mark Podlipec (podlipec@ici.net). For those pieces of code,
* the following copyrights notice applies:
*
* XAnim Copyright (C) 1990-1997 by Mark Podlipec.
* All rights reserved.
*
* This software may be freely copied, modified and redistributed without
* fee for non-commerical purposes provided that this copyright notice is
* preserved intact on all copies and modified copies.
*
* There is no warranty or other guarantee of fitness of this software.
* It is provided solely "as is". The author(s) disclaim(s) all
* responsibility and liability with respect to this software's usage
* or its effect upon hardware or computer systems.
*
* 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 */
#ifdef HAVE_MALLOC_H
#include <malloc.h>
#endif
#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"
/* Private data for .wav file */
typedef struct wavstuff {
LONG numSamples;
int second_header; /* non-zero on second header write */
unsigned short formatTag; /* What type of encoding file is using */
/* The following are only needed for ADPCM wav files */
unsigned short samplesPerBlock;
unsigned short bytesPerBlock;
unsigned short blockAlign;
short *samples[2]; /* Left and Right sample buffers */
short *samplePtr[2]; /* Pointers to current samples */
unsigned short blockSamplesRemaining;/* Samples remaining in each channel */
unsigned char *packet; /* Temporary buffer for packets */
} *wav_t;
static char *wav_format_str();
void wavwritehdr(P1(ft_t));
/*
*
* Lookup tables for MS ADPCM format
*
*/
static LONG gaiP4[] = { 230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230 };
/* TODO : The first 7 coef's are are always hardcode and must
appear in the actual WAVE file. They should be read in
in case a sound program added extras to the list. */
static LONG gaiCoef1[] = { 256, 512, 0, 192, 240, 460, 392 };
static LONG gaiCoef2[] = { 0, -256, 0, 64, 0,-208, -232};
/*
*
* Lookup tables for IMA ADPCM format
*
*/
static int imaIndexAdjustTable[16] = {
-1, -1, -1, -1, /* +0 - +3, decrease the step size */
2, 4, 6, 8, /* +4 - +7, increase the step size */
-1, -1, -1, -1, /* -0 - -3, decrease the step size */
2, 4, 6, 8, /* -4 - -7, increase the step size */
};
static int imaStepSizeTable[89] = {
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34,
37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494,
544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552,
1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026,
4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442,
11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623,
27086, 29794, 32767
};
/****************************************************************************/
/* IMA ADPCM Support Functions Section */
/****************************************************************************/
/*
*
* MsAdpcmDecode - Decode a given sample and update state tables
*
*/
short ImaAdpcmDecode(deltaCode, state)
unsigned char deltaCode;
ImaState_t *state;
{
/* Get the current step size */
int step;
int difference;
step = imaStepSizeTable[state->index];
/* Construct the difference by scaling the current step size */
/* This is approximately: difference = (deltaCode+.5)*step/4 */
difference = step>>3;
if ( deltaCode & 1 ) difference += step>>2;
if ( deltaCode & 2 ) difference += step>>1;
if ( deltaCode & 4 ) difference += step;
if ( deltaCode & 8 ) difference = -difference;
/* Build the new sample */
state->previousValue += difference;
if (state->previousValue > 32767) state->previousValue = 32767;
else if (state->previousValue < -32768) state->previousValue = -32768;
/* Update the step for the next sample */
state->index += imaIndexAdjustTable[deltaCode];
if (state->index < 0) state->index = 0;
else if (state->index > 88) state->index = 88;
return state->previousValue;
}
/*
*
* ImaAdpcmNextBlock - Grab and decode complete block of samples
*
*/
unsigned short ImaAdpcmNextBlock(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
/* Pull in the packet and check the header */
unsigned short bytesRead;
unsigned char *bytePtr;
ImaState_t state[2]; /* One decompressor state for each channel */
int ch;
unsigned short remaining;
unsigned short samplesThisBlock;
int i;
unsigned char b;
bytesRead = fread(wav->packet,1,wav->blockAlign,ft->fp);
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 then trailing quite. */
if (bytesRead >= (4 * ft->info.channels))
{
samplesThisBlock = (wav->blockAlign - (3 * ft->info.channels));
}
else
{
warn ("Premature EOF on .wav input file");
return 0;
}
}
else
samplesThisBlock = wav->samplesPerBlock;
bytePtr = wav->packet;
/* Read the four-byte header for each channel */
/* Reset the decompressor */
for(ch=0;ch < ft->info.channels; ch++) {
/* Got this from xanim */
state[ch].previousValue = ((int)bytePtr[1]<<8) +
(int)bytePtr[0];
if (state[ch].previousValue & 0x8000)
state[ch].previousValue -= 0x10000;
if (bytePtr[2] > 88)
{
warn("IMA ADPCM Format Error (bad index value) in wav file");
state[ch].index = 88;
}
else
state[ch].index = bytePtr[2];
if (bytePtr[3])
warn("IMA ADPCM Format Error (synchronization error) in wav file");
bytePtr+=4; /* Skip this header */
wav->samplePtr[ch] = wav->samples[ch];
/* Decode one sample for the header */
*(wav->samplePtr[ch]++) = state[ch].previousValue;
}
/* Decompress nybbles. Remainging is bytes in block minus header */
/* Subtract the one sample taken from header */
remaining = samplesThisBlock-1;
while (remaining) {
/* Always decode 8 samples */
remaining -= 8;
/* Decode 8 left samples */
for (i=0;i<4;i++) {
b = *bytePtr++;
*(wav->samplePtr[0]++) = ImaAdpcmDecode(b & 0x0f,&state[0]);
*(wav->samplePtr[0]++) = ImaAdpcmDecode((b>>4) & 0x0f,&state[0]);
}
if (ft->info.channels < 2)
continue; /* If mono, skip rest of loop */
/* Decode 8 right samples */
for (i=0;i<4;i++) {
b = *bytePtr++;
*(wav->samplePtr[1]++) = ImaAdpcmDecode(b & 0x0f,&state[1]);
*(wav->samplePtr[1]++) = ImaAdpcmDecode((b>>4) & 0x0f,&state[1]);
}
}
/* For a full block, the following should be true: */
/* wav->samplesPerBlock = blockAlign - 8byte header + 1 sample in header */
return wav->samplesPerBlock;
}
/****************************************************************************/
/* MS ADPCM Support Functions Section */
/****************************************************************************/
/*
*
* MsAdpcmDecode - Decode a given sample and update state tables
*
*/
LONG MsAdpcmDecode(deltaCode, state)
LONG deltaCode;
MsState_t *state;
{
LONG predict;
LONG sample;
LONG idelta;
/** Compute next Adaptive Scale Factor (ASF) **/
idelta = state->index;
state->index = (gaiP4[deltaCode] * idelta) >> 8;
if (state->index < 16) state->index = 16;
if (deltaCode & 0x08) deltaCode = deltaCode - 0x10;
/** Predict next sample **/
predict = ((state->sample1 * gaiCoef1[state->bpred]) + (state->sample2 * gaiCoef2[state->bpred])) >> 8;
/** reconstruct original PCM **/
sample = (deltaCode * idelta) + predict;
if (sample > 32767) sample = 32767;
else if (sample < -32768) sample = -32768;
state->sample2 = state->sample1;
state->sample1 = sample;
return (sample);
}
/*
*
* MsAdpcmNextBlock - Grab and decode complete block of samples
*
*/
unsigned short MsAdpcmNextBlock(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
unsigned short bytesRead;
unsigned char *bytePtr;
MsState_t state[2]; /* One decompressor state for each channel */
unsigned short samplesThisBlock;
unsigned short remaining;
unsigned char b;
/* Pull in the packet and check the header */
bytesRead = fread(wav->packet,1,wav->blockAlign,ft->fp);
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 then trailing quite. */
if (bytesRead >= (7 * ft->info.channels))
{
samplesThisBlock = (wav->blockAlign - (6 * ft->info.channels));
}
else
{
warn ("Premature EOF on .wav input file");
return 0;
}
}
else
samplesThisBlock = wav->samplesPerBlock;
bytePtr = wav->packet;
/* Read the four-byte header for each channel */
/* Reset the decompressor */
state[0].bpred = *bytePtr++; /* Left */
if (ft->info.channels > 1)
state[1].bpred = *bytePtr++; /* Right */
else
state[1].bpred = 0;
/* 7 should be variable from AVI/WAV header */
if (state[0].bpred >= 7)
{
warn("MSADPCM bpred %x and should be less than 7\n",state[0].bpred);
return(0);
}
if (state[1].bpred >= 7)
{
warn("MSADPCM bpred %x and should be less than 7\n",state[1].bpred);
return(0);
}
state[0].index = *bytePtr++; state[0].index |= (*bytePtr++)<<8;
if (state[0].index & 0x8000) state[0].index -= 0x10000;
if (ft->info.channels > 1)
{
state[1].index = *bytePtr++; state[1].index |= (*bytePtr++)<<8;
if (state[1].index & 0x8000) state[1].index -= 0x10000;
}
state[0].sample1 = *bytePtr++; state[0].sample1 |= (*bytePtr++)<<8;
if (state[0].sample1 & 0x8000) state[0].sample1 -= 0x10000;
if (ft->info.channels > 1)
{
state[1].sample1 = *bytePtr++; state[1].sample1 |= (*bytePtr++)<<8;
if (state[1].sample1 & 0x8000) state[1].sample1 -= 0x10000;
}
state[0].sample2 = *bytePtr++; state[0].sample2 |= (*bytePtr++)<<8;
if (state[0].sample2 & 0x8000) state[0].sample2 -= 0x10000;
if (ft->info.channels > 1)
{
state[1].sample2 = *bytePtr++; state[1].sample2 |= (*bytePtr++)<<8;
if (state[1].sample2 & 0x8000) state[1].sample2 -= 0x10000;
}
wav->samplePtr[0] = wav->samples[0];
wav->samplePtr[1] = wav->samples[1];
/* Decode two samples for the header */
*(wav->samplePtr[0]++) = state[0].sample2;
*(wav->samplePtr[0]++) = state[0].sample1;
if (ft->info.channels > 1)
{
*(wav->samplePtr[1]++) = state[1].sample2;
*(wav->samplePtr[1]++) = state[1].sample1;
}
/* Decompress nybbles. Minus 2 included in header */
remaining = samplesThisBlock-2;
while (remaining) {
b = *bytePtr++;
*(wav->samplePtr[0]++) = MsAdpcmDecode((b>>4) & 0x0f, &state[0]);
remaining--;
if (ft->info.channels == 1)
{
*(wav->samplePtr[0]++) = MsAdpcmDecode(b & 0x0f, &state[0]);
remaining--;
}
else
{
*(wav->samplePtr[1]++) = MsAdpcmDecode(b & 0x0f, &state[1]);
}
}
return samplesThisBlock;
}
/****************************************************************************/
/* General Sox WAV file code */
/****************************************************************************/
/*
* Do anything required before you start reading samples.
* Read file header.
* Find out sampling rate,
* size and style of samples,
* mono/stereo/quad.
*/
void wavstartread(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
char magic[4];
ULONG len;
int littlendian = 1;
char *endptr;
/* wave file characteristics */
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 wExtSize = 0; /* extended field for ADPCM */
unsigned short wNumCoefs = 0; /* Related to IMA ADPCM */
ULONG data_length; /* length of sound data in bytes */
ULONG bytespersample; /* bytes per sample (per channel */
/* This is needed for rawread() */
rawstartread(ft);
endptr = (char *) &littlendian;
if (!*endptr) ft->swap = ft->swap ? 0 : 1;
/* If you need to seek around the input file. */
if (0 && ! ft->seekable)
fail("WAVE input file must be a file, not a pipe");
if ( fread(magic, 1, 4, ft->fp) != 4 || strncmp("RIFF", magic, 4))
fail("WAVE: RIFF header not found");
len = rlong(ft);
if ( fread(magic, 1, 4, ft->fp) != 4 || strncmp("WAVE", magic, 4))
fail("WAVE header not found");
/* Now look for the format chunk */
for (;;)
{
if ( fread(magic, 1, 4, ft->fp) != 4 )
fail("WAVE file missing fmt spec");
len = rlong(ft);
if (strncmp("fmt ", magic, 4) == 0)
break; /* Found the format chunk */
/* skip to next chunk */
while (len > 0 && !feof(ft->fp))
{
getc(ft->fp);
len--;
}
}
if ( len < 16 )
fail("WAVE file fmt chunk is too short");
wav->formatTag = rshort(ft);
len -= 2;
switch (wav->formatTag)
{
case WAVE_FORMAT_UNKNOWN:
fail("WAVE file is in unsupported Microsoft Official Unknown format.");
case WAVE_FORMAT_PCM:
/* Default (-1) depends on sample size. Set that later on. */
if (ft->info.style != -1 && ft->info.style != UNSIGNED &&
ft->info.style != SIGN2)
warn("User options overriding style read in .wav header");
break;
case WAVE_FORMAT_ADPCM:
case WAVE_FORMAT_IMA_ADPCM:
if (ft->info.style == -1 || ft->info.style == ADPCM)
ft->info.style = ADPCM;
else
warn("User options overriding style read in .wav header");
break;
case WAVE_FORMAT_IEEE_FLOAT:
fail("Sorry, this WAV file is in IEEE Float format.");
case WAVE_FORMAT_ALAW:
if (ft->info.style == -1 || ft->info.style == ALAW)
ft->info.style = ALAW;
else
warn("User options overriding style read in .wav header");
break;
case WAVE_FORMAT_MULAW:
if (ft->info.style == -1 || ft->info.style == ULAW)
ft->info.style = ULAW;
else
warn("User options overriding style read in .wav header");
break;
case WAVE_FORMAT_OKI_ADPCM:
fail("Sorry, this WAV file is in OKI ADPCM format.");
case WAVE_FORMAT_DIGISTD:
fail("Sorry, this WAV file is in Digistd format.");
case WAVE_FORMAT_DIGIFIX:
fail("Sorry, this WAV file is in Digifix format.");
case WAVE_FORMAT_DOLBY_AC2:
fail("Sorry, this WAV file is in Dolby AC2 format.");
case WAVE_FORMAT_GSM610:
fail("Sorry, this WAV file is in GSM 6.10 format.");
case WAVE_FORMAT_ROCKWELL_ADPCM:
fail("Sorry, this WAV file is in Rockwell ADPCM format.");
case WAVE_FORMAT_ROCKWELL_DIGITALK:
fail("Sorry, this WAV file is in Rockwell DIGITALK format.");
case WAVE_FORMAT_G721_ADPCM:
fail("Sorry, this WAV file is in G.721 ADPCM format.");
case WAVE_FORMAT_G728_CELP:
fail("Sorry, this WAV file is in G.728 CELP format.");
case WAVE_FORMAT_MPEG:
fail("Sorry, this WAV file is in MPEG format.");
case WAVE_FORMAT_MPEGLAYER3:
fail("Sorry, this WAV file is in MPEG Layer 3 format.");
case WAVE_FORMAT_G726_ADPCM:
fail("Sorry, this WAV file is in G.726 ADPCM format.");
case WAVE_FORMAT_G722_ADPCM:
fail("Sorry, this WAV file is in G.722 ADPCM format.");
default: fail("WAV file has unknown format type of %x",wav->formatTag);
}
wChannels = rshort(ft);
len -= 2;
/* User options take precedence */
if (ft->info.channels == -1 || ft->info.channels == wChannels)
ft->info.channels = wChannels;
else
warn("User options overriding channels read in .wav header");
wSamplesPerSecond = rlong(ft);
len -= 4;
if (ft->info.rate == 0 || ft->info.rate == wSamplesPerSecond)
ft->info.rate = wSamplesPerSecond;
else
warn("User options overriding rate read in .wav header");
wAvgBytesPerSec = rlong(ft); /* Average bytes/second */
wav->blockAlign = rshort(ft); /* Block align */
len -= 6;
/* bits per sample per channel */
wBitsPerSample = rshort(ft);
len -= 2;
/* ADPCM formats have extended fmt chunk. Check for those cases. */
if (wav->formatTag == WAVE_FORMAT_ADPCM)
{
if (wBitsPerSample != 4)
fail("Can only handle 4-bit MS ADPCM in wav files");
wExtSize = rshort(ft);
wav->samplesPerBlock = rshort(ft);
wav->bytesPerBlock = (wav->samplesPerBlock + 7)/2 * ft->info.channels;
wNumCoefs = rshort(ft);
wav->packet = (unsigned char *)malloc(wav->blockAlign);
len -= 6;
wav->samples[1] = wav->samples[0] = 0;
/* Use ft->info.channels after this becuase wChannels is now bad */
while (wChannels-- > 0)
wav->samples[wChannels] = (short *)malloc(wav->samplesPerBlock*sizeof(short));
/* Here we are setting the bytespersample AFTER de-compression */
bytespersample = WORD;
}
else if (wav->formatTag == WAVE_FORMAT_IMA_ADPCM)
{
if (wBitsPerSample != 4)
fail("Can only handle 4-bit IMA ADPCM in wav files");
wExtSize = rshort(ft);
wav->samplesPerBlock = rshort(ft);
wav->bytesPerBlock = (wav->samplesPerBlock + 7)/2 * ft->info.channels;
wav->packet = (unsigned char *)malloc(wav->blockAlign);
len -= 4;
wav->samples[1] = wav->samples[0] = 0;
/* Use ft->info.channels after this becuase wChannels is now bad */
while (wChannels-- > 0)
wav->samples[wChannels] = (short *)malloc(wav->samplesPerBlock*sizeof(short));
/* Here we are setting the bytespersample AFTER de-compression */
bytespersample = WORD;
}
else
{
bytespersample = (wBitsPerSample + 7)/8;
}
switch (bytespersample)
{
case BYTE:
/* User options take precedence */
if (ft->info.size == -1 || ft->info.size == BYTE)
ft->info.size = BYTE;
else
warn("User options overriding size read in .wav header");
/* Now we have enough information to set default styles. */
if (ft->info.style == -1)
ft->info.style = UNSIGNED;
break;
case WORD:
if (ft->info.size == -1 || ft->info.size == WORD)
ft->info.size = WORD;
else
warn("User options overriding size read in .wav header");
/* Now we have enough information to set default styles. */
if (ft->info.style == -1)
ft->info.style = SIGN2;
break;
case DWORD:
if (ft->info.size == -1 || ft->info.size == DWORD)
ft->info.size = DWORD;
else
warn("User options overriding size read in .wav header");
/* Now we have enough information to set default styles. */
if (ft->info.style == -1)
ft->info.style = SIGN2;
break;
default:
fail("Sorry, don't understand .wav size");
}
/* Skip past the rest of any left over fmt chunk */
while (len > 0 && !feof(ft->fp))
{
getc(ft->fp);
len--;
}
/* Now look for the wave data chunk */
for (;;)
{
if ( fread(magic, 1, 4, ft->fp) != 4 )
fail("WAVE file has missing data chunk");
len = rlong(ft);
if (strncmp("data", magic, 4) == 0)
break; /* Found the data chunk */
while (len > 0 && !feof(ft->fp)) /* skip to next chunk */
{
getc(ft->fp);
len--;
}
}
data_length = len;
if (wav->formatTag == WAVE_FORMAT_ADPCM)
{
/* Compute easiest part of number of samples. For every block, there
are samplesPerBlock samples to read. */
wav->numSamples = (((data_length / wav->blockAlign) * wav->samplesPerBlock) * ft->info.channels);
/* Next, for any partial blocks, substract overhead from it and it
will leave # of samples to read. */
wav->numSamples += ((data_length - ((data_length/wav->blockAlign)
*wav->blockAlign))
- (6 * ft->info.channels)) * ft->info.channels;
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
}
else if (wav->formatTag == WAVE_FORMAT_IMA_ADPCM)
{
/* Compute easiest part of number of samples. For every block, there
are samplesPerBlock samples to read. */
wav->numSamples = (((data_length / wav->blockAlign) * wav->samplesPerBlock) * ft->info.channels);
/* Next, for any partial blocks, substract overhead from it and it
will leave # of samples to read. */
wav->numSamples += ((data_length - ((data_length/wav->blockAlign)
*wav->blockAlign))
- (3 * ft->info.channels)) * ft->info.channels;
wav->blockSamplesRemaining = 0; /* Samples left in buffer */
}
else
wav->numSamples = data_length/ft->info.size; /* total samples */
report("Reading Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wav->formatTag), ft->info.channels,
wChannels == 1 ? "" : "s", wSamplesPerSecond);
report(" %d byte/sec, %d block align, %d bits/samp, %u data bytes",
wAvgBytesPerSec, wav->blockAlign, wBitsPerSample, data_length);
/* Can also report exteded fmt information */
if (wav->formatTag == WAVE_FORMAT_ADPCM)
report(" %d Extsize, %d Samps/block, %d bytes/block %d Num Coefs\n",wExtSize,wav->samplesPerBlock,wav->bytesPerBlock,wNumCoefs);
else if (wav->formatTag == WAVE_FORMAT_IMA_ADPCM)
report(" %d Extsize, %d Samps/block, %d bytes/block\n",wExtSize,wav->samplesPerBlock,wav->bytesPerBlock);
}
/*
* Read up to len samples from file.
* Convert to signed longs.
* Place in buf[].
* Return number of samples read.
*/
LONG wavread(ft, buf, len)
ft_t ft;
LONG *buf, len;
{
wav_t wav = (wav_t) ft->priv;
LONG done;
if (len > wav->numSamples) len = wav->numSamples;
/* If file is in ADPCM style then read in multiple blocks else */
/* read as much as possible and return quickly. */
if (ft->info.style == ADPCM)
{
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 = ImaAdpcmNextBlock(ft);
else
wav->blockSamplesRemaining = MsAdpcmNextBlock(ft);
if (wav->blockSamplesRemaining == 0)
{
/* Don't try to read any more samples */
wav->numSamples = 0;
return done;
}
wav->samplePtr[0] = wav->samples[0];
wav->samplePtr[1] = wav->samples[1];
}
switch(ft->info.channels) { /* Copy data into buf */
case 1: /* Mono: Just copy left channel data */
while ((wav->blockSamplesRemaining > 0) && (done < len))
{
/* Output is already signed */
*buf++ = LEFT(*(wav->samplePtr[0]++), 16);
done++;
wav->blockSamplesRemaining--;
}
break;
case 2: /* Stereo: Interleave samples */
while ((wav->blockSamplesRemaining > 0) && (done < len))
{
/* Output is already signed */
*buf++ = LEFT(*(wav->samplePtr[0]++),16); /* Left */
*buf++ = LEFT(*(wav->samplePtr[1]++),16); /* Right */
done += 2;
wav->blockSamplesRemaining--;
}
break;
default:
fail ("Can only handle stereo or mono files");
}
}
}
else /* else not ADPCM style */
{
done = rawread(ft, buf, len);
/* If software thinks there are more samples but I/O */
/* says otherwise, let the user no about this. */
if (done == 0 && wav->numSamples != 0)
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!
*/
void wavstopread(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
if (wav->packet) free(wav->packet);
if (wav->samples[0]) free(wav->samples[0]);
if (wav->samples[1]) free(wav->samples[1]);
/* Needed for rawread() */
rawstopread(ft);
}
void wavstartwrite(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
int littlendian = 1;
char *endptr;
endptr = (char *) &littlendian;
if (!*endptr) ft->swap = ft->swap ? 0 : 1;
wav->numSamples = 0;
wav->second_header = 0;
if (! ft->seekable)
warn("Length in output .wav header will wrong since can't seek to fix it");
wavwritehdr(ft);
}
void wavwritehdr(ft)
ft_t ft;
{
wav_t wav = (wav_t) ft->priv;
/* wave file characteristics */
unsigned short wFormatTag = 0; /* data format */
unsigned short wChannels; /* number of channels */
ULONG wSamplesPerSecond; /* samples per second per channel */
ULONG wAvgBytesPerSec; /* estimate of bytes per second needed */
unsigned short wBlockAlign; /* byte alignment of a basic sample block */
unsigned short wBitsPerSample; /* bits per sample */
ULONG data_length; /* length of sound data in bytes */
ULONG bytespersample; /* bytes per sample (per channel) */
/* Needed for rawwrite() */
rawstartwrite(ft);
switch (ft->info.size)
{
case BYTE:
wBitsPerSample = 8;
if (ft->info.style != UNSIGNED &&
ft->info.style != ULAW &&
ft->info.style != ALAW &&
!wav->second_header)
{
warn("Only support unsigned, ulaw, or alaw with 8-bit data. Forcing to unsigned");
ft->info.style = UNSIGNED;
}
break;
case WORD:
wBitsPerSample = 16;
if ((ft->info.style == UNSIGNED ||
ft->info.style == ULAW ||
ft->info.style == ALAW) &&
!wav->second_header)
{
warn("Do not support Unsigned, ulaw, or alaw with 16 bit data. Forcing to Signed");
ft->info.style = SIGN2;
}
break;
case DWORD:
wBitsPerSample = 32;
break;
default:
wBitsPerSample = 32;
break;
}
switch (ft->info.style)
{
case UNSIGNED:
wFormatTag = WAVE_FORMAT_PCM;
break;
case SIGN2:
wFormatTag = WAVE_FORMAT_PCM;
break;
case ALAW:
wFormatTag = WAVE_FORMAT_ALAW;
break;
case ULAW:
wFormatTag = WAVE_FORMAT_MULAW;
break;
case ADPCM:
wFormatTag = WAVE_FORMAT_PCM;
warn("Can not support writing ADPCM style. Overriding to Signed Words\n");
ft->info.style = SIGN2;
wBitsPerSample = 16;
/* wFormatTag = WAVE_FORMAT_IMA_ADPCM;
wBitsPerSample = 4;
if (wBitsPerSample != 4 && !wav->second_header)
break; */
}
wSamplesPerSecond = ft->info.rate;
bytespersample = (wBitsPerSample + 7)/8;
wAvgBytesPerSec = ft->info.rate * ft->info.channels * bytespersample;
wChannels = ft->info.channels;
wBlockAlign = ft->info.channels * bytespersample;
if (!wav->second_header) /* use max length value first time */
data_length = 0x7fffffffL - (8+16+12);
else /* fixup with real length */
{
if (ft->info.style == ADPCM)
data_length = wav->numSamples / 2;
else
data_length = bytespersample * wav->numSamples;
}
/* figured out header info, so write it */
fputs("RIFF", ft->fp);
wlong(ft, data_length + 8+16+12); /* Waveform chunk size: FIXUP(4) */
fputs("WAVE", ft->fp);
fputs("fmt ", ft->fp);
wlong(ft, (LONG)16); /* fmt chunk size */
wshort(ft, wFormatTag);
wshort(ft, wChannels);
wlong(ft, wSamplesPerSecond);
wlong(ft, wAvgBytesPerSec);
wshort(ft, wBlockAlign);
wshort(ft, wBitsPerSample);
fputs("data", ft->fp);
wlong(ft, data_length); /* data chunk size: FIXUP(40) */
if (!wav->second_header) {
report("Writing Wave file: %s format, %d channel%s, %d samp/sec",
wav_format_str(wFormatTag), wChannels,
wChannels == 1 ? "" : "s", wSamplesPerSecond);
report(" %d byte/sec, %d block align, %d bits/samp",
wAvgBytesPerSec, wBlockAlign, wBitsPerSample);
} else
report("Finished writing Wave file, %u data bytes\n",data_length);
}
void wavwrite(ft, buf, len)
ft_t ft;
LONG *buf, len;
{
wav_t wav = (wav_t) ft->priv;
wav->numSamples += len;
rawwrite(ft, buf, len);
}
void
wavstopwrite(ft)
ft_t ft;
{
/* Call this to flush out any remaining data. */
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;
if (fseek(ft->fp, 0L, SEEK_SET) != 0)
fail("Sorry, can't rewind output file to rewrite .wav header.");
((wav_t) ft->priv)->second_header = 1;
wavwritehdr(ft);
}
/*
* 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";
}
}