ref: 5d8df0e043f05d080a990d32cfca46e00758953a
dir: /src/smploaders/pt2_load_aiff.c/
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <stdbool.h>
#include <math.h>
#include "../pt2_header.h"
#include "../pt2_config.h"
#include "../pt2_structs.h"
#include "../pt2_textout.h"
#include "../pt2_visuals.h"
#include "../pt2_helpers.h"
#include "../pt2_replayer.h"
#include "../pt2_askbox.h"
#include "../pt2_downsample2x.h"
#include "../pt2_audio.h"
static uint32_t getAIFFSampleRate(uint8_t *in)
{
/* 80-bit IEEE-754 to unsigned 32-bit integer (rounded).
** Sign bit is ignored.
*/
#define EXP_BIAS 16383
const uint16_t exp15 = ((in[0] & 0x7F) << 8) | in[1];
const uint64_t mantissaBits = *(uint64_t *)&in[2];
const uint64_t mantissa63 = SWAP64(mantissaBits) & INT64_MAX;
double dExp = exp15 - EXP_BIAS;
double dMantissa = mantissa63 / (INT64_MAX+1.0);
double dResult = (1.0 + dMantissa) * exp2(dExp);
return (uint32_t)round(dResult);
}
bool loadAIFFSample(FILE *f, uint32_t filesize, moduleSample_t *s)
{
uint8_t sampleRateBytes[10];
uint16_t bitDepth, numChannels;
uint32_t offset, sampleRate;
// zero out chunk pointers and lengths
uint32_t commPtr = 0; uint32_t commLen = 0;
uint32_t ssndPtr = 0; uint32_t ssndLen = 0;
bool unsigned8bit = false;
fseek(f, 12, SEEK_SET);
while (!feof(f) && (uint32_t)ftell(f) < filesize-12)
{
uint32_t blockName, blockSize;
fread(&blockName, 4, 1, f); if (feof(f)) break;
fread(&blockSize, 4, 1, f); if (feof(f)) break;
blockName = SWAP32(blockName);
blockSize = SWAP32(blockSize);
switch (blockName)
{
case 0x434F4D4D: // "COMM"
{
commPtr = ftell(f);
commLen = blockSize;
}
break;
case 0x53534E44: // "SSND"
{
ssndPtr = ftell(f);
ssndLen = blockSize;
}
break;
default: break;
}
fseek(f, blockSize + (blockSize & 1), SEEK_CUR);
}
if (commPtr == 0 || commLen < 18 || ssndPtr == 0)
{
displayErrorMsg("NOT A VALID AIFF!");
return false;
}
// kludge for some really strange AIFFs
if (ssndLen == 0)
ssndLen = filesize - ssndPtr;
if (ssndPtr+ssndLen > (uint32_t)filesize)
ssndLen = filesize - ssndPtr;
fseek(f, commPtr, SEEK_SET);
fread(&numChannels, 2, 1, f); numChannels = SWAP16(numChannels);
fseek(f, 4, SEEK_CUR);
fread(&bitDepth, 2, 1, f); bitDepth = SWAP16(bitDepth);
fread(sampleRateBytes, 1, 10, f);
fseek(f, 4 + 2 + 1, SEEK_CUR);
if (numChannels != 1 && numChannels != 2) // sample type
{
displayErrorMsg("UNSUPPORTED AIFF!");
return false;
}
if (bitDepth != 8 && bitDepth != 16 && bitDepth != 24 && bitDepth != 32)
{
displayErrorMsg("UNSUPPORTED AIFF!");
return false;
}
// read compression type (if present)
if (commLen > 18)
{
char compType[4];
fread(compType, 1, 4, f);
if (memcmp(compType, "NONE", 4))
{
displayErrorMsg("UNSUPPORTED AIFF!");
return false;
}
}
sampleRate = getAIFFSampleRate(sampleRateBytes);
// sample data chunk
fseek(f, ssndPtr, SEEK_SET);
fread(&offset, 4, 1, f);
if (offset > 0)
{
displayErrorMsg("UNSUPPORTED AIFF!");
return false;
}
fseek(f, 4, SEEK_CUR);
ssndLen -= 8; // don't include offset and blockSize datas
int32_t sampleLength = ssndLen;
if (sampleLength == 0)
{
displayErrorMsg("NOT A VALID AIFF!");
return false;
}
bool downSample = false;
if (sampleRate > 22050 && !config.noDownsampleOnSmpLoad)
{
if (askBox(ASKBOX_DOWNSAMPLE, "DOWNSAMPLE ?"))
downSample = true;
}
int8_t *smpDataPtr = &song->sampleData[s->offset];
if (bitDepth == 8) // 8-BIT INTEGER SAMPLE
{
if (sampleLength > config.maxSampleLength*4)
sampleLength = config.maxSampleLength*4;
int8_t *audioDataS8 = (int8_t *)malloc(sampleLength * sizeof (int8_t));
if (audioDataS8 == NULL)
{
statusOutOfMemory();
return false;
}
// read sample data
if (fread(audioDataS8, 1, sampleLength, f) != (size_t)sampleLength)
{
free(audioDataS8);
displayErrorMsg("I/O ERROR !");
return false;
}
if (unsigned8bit)
{
for (int32_t i = 0; i < sampleLength; i++)
audioDataS8[i] ^= 0x80;
}
// convert from stereo to mono (if needed)
if (numChannels == 2)
{
sampleLength >>= 1;
for (int32_t i = 0; i < sampleLength-1; i++) // add right channel to left channel
audioDataS8[i] = (audioDataS8[(i * 2) + 0] + audioDataS8[(i * 2) + 1]) >> 1;;
}
// 2x downsampling
if (downSample)
{
downsample2x8Bit(audioDataS8, sampleLength);
sampleLength >>= 1;
}
if (sampleLength > config.maxSampleLength)
sampleLength = config.maxSampleLength;
turnOffVoices();
for (int32_t i = 0; i < sampleLength; i++)
smpDataPtr[i] = audioDataS8[i];
free(audioDataS8);
}
else if (bitDepth == 16) // 16-BIT INTEGER SAMPLE
{
sampleLength >>= 1;
if (sampleLength > config.maxSampleLength*4)
sampleLength = config.maxSampleLength*4;
int16_t *audioDataS16 = (int16_t *)malloc(sampleLength * sizeof (int16_t));
if (audioDataS16 == NULL)
{
statusOutOfMemory();
return false;
}
// read sample data
if (fread(audioDataS16, 2, sampleLength, f) != (size_t)sampleLength)
{
free(audioDataS16);
displayErrorMsg("I/O ERROR !");
return false;
}
// fix endianness
for (int32_t i = 0; i < sampleLength; i++)
audioDataS16[i] = SWAP16(audioDataS16[i]);
// convert from stereo to mono (if needed)
if (numChannels == 2)
{
sampleLength >>= 1;
for (int32_t i = 0; i < sampleLength-1; i++) // add right channel to left channel
audioDataS16[i] = (audioDataS16[(i << 1) + 0] + audioDataS16[(i << 1) + 1]) >> 1;
}
// 2x downsampling
if (downSample)
{
downsample2x16Bit(audioDataS16, sampleLength);
sampleLength >>= 1;
}
if (sampleLength > config.maxSampleLength)
sampleLength = config.maxSampleLength;
double dAmp = 1.0;
if (downSample) // we already normalized
{
dAmp = INT8_MAX / (double)INT16_MAX;
}
else
{
const double dPeak = get16BitPeak(audioDataS16, sampleLength);
if (dPeak > 0.0)
dAmp = INT8_MAX / dPeak;
}
turnOffVoices();
for (int32_t i = 0; i < sampleLength; i++)
{
int32_t smp32 = (int32_t)round(audioDataS16[i] * dAmp);
assert(smp32 >= -128 && smp32 <= 127); // shouldn't happen according to dAmp (but just in case)
smpDataPtr[i] = (int8_t)smp32;
}
free(audioDataS16);
}
else if (bitDepth == 24) // 24-BIT INTEGER SAMPLE
{
sampleLength /= 3;
if (sampleLength > config.maxSampleLength*4)
sampleLength = config.maxSampleLength*4;
int32_t *audioDataS32 = (int32_t *)malloc(sampleLength * sizeof (int32_t));
if (audioDataS32 == NULL)
{
statusOutOfMemory();
return false;
}
// read sample data
if (fread(&audioDataS32[sampleLength >> 2], 3, sampleLength, f) != (size_t)sampleLength)
{
free(audioDataS32);
displayErrorMsg("I/O ERROR !");
return false;
}
// convert to 32-bit
uint8_t *audioDataU8 = (uint8_t *)audioDataS32 + sampleLength;
for (int32_t i = 0; i < sampleLength; i++)
{
audioDataS32[i] = (audioDataU8[0] << 24) | (audioDataU8[1] << 16) | (audioDataU8[2] << 8);
audioDataU8 += 3;
}
// convert from stereo to mono (if needed)
if (numChannels == 2)
{
sampleLength >>= 1;
for (int32_t i = 0; i < sampleLength-1; i++) // add right channel to left channel
{
int64_t smp = ((int64_t)audioDataS32[(i << 1) + 0] + audioDataS32[(i << 1) + 1]) >> 1;
audioDataS32[i] = (int32_t)smp;
}
}
// 2x downsampling
if (downSample)
{
downsample2x32Bit(audioDataS32, sampleLength);
sampleLength >>= 1;
}
if (sampleLength > config.maxSampleLength)
sampleLength = config.maxSampleLength;
double dAmp = 1.0;
if (downSample) // we already normalized
{
dAmp = INT8_MAX / (double)INT32_MAX;
}
else
{
const double dPeak = get32BitPeak(audioDataS32, sampleLength);
if (dPeak > 0.0)
dAmp = INT8_MAX / dPeak;
}
turnOffVoices();
for (int32_t i = 0; i < sampleLength; i++)
{
int32_t smp32 = (int32_t)round(audioDataS32[i] * dAmp);
smpDataPtr[i] = (int8_t)smp32;
}
free(audioDataS32);
}
else if (bitDepth == 32) // 32-BIT INTEGER SAMPLE
{
sampleLength >>= 2;
if (sampleLength > config.maxSampleLength*4)
sampleLength = config.maxSampleLength*4;
int32_t *audioDataS32 = (int32_t *)malloc(sampleLength * sizeof (int32_t));
if (audioDataS32 == NULL)
{
statusOutOfMemory();
return false;
}
// read sample data
if (fread(audioDataS32, 4, sampleLength, f) != (size_t)sampleLength)
{
free(audioDataS32);
displayErrorMsg("I/O ERROR !");
return false;
}
// fix endianness
for (int32_t i = 0; i < sampleLength; i++)
audioDataS32[i] = SWAP32(audioDataS32[i]);
// convert from stereo to mono (if needed)
if (numChannels == 2)
{
sampleLength >>= 1;
for (int32_t i = 0; i < sampleLength-1; i++) // add right channel to left channel
{
int64_t smp = ((int64_t)audioDataS32[(i << 1) + 0] + audioDataS32[(i << 1) + 1]) >> 1;
audioDataS32[i] = (int32_t)smp;
}
}
// 2x downsampling
if (downSample)
{
downsample2x32Bit(audioDataS32, sampleLength);
sampleLength >>= 1;
}
if (sampleLength > config.maxSampleLength)
sampleLength = config.maxSampleLength;
double dAmp = 1.0;
if (downSample) // we already normalized
{
dAmp = INT8_MAX / (double)INT32_MAX;
}
else
{
const double dPeak = get32BitPeak(audioDataS32, sampleLength);
if (dPeak > 0.0)
dAmp = INT8_MAX / dPeak;
}
turnOffVoices();
for (int32_t i = 0; i < sampleLength; i++)
{
int32_t smp32 = (int32_t)round(audioDataS32[i] * dAmp);
smpDataPtr[i] = (int8_t)smp32;
}
free(audioDataS32);
}
if (sampleLength & 1)
{
if (++sampleLength > config.maxSampleLength)
sampleLength = config.maxSampleLength;
}
s->length = sampleLength;
s->fineTune = 0;
s->volume = 64;
s->loopStart = 0;
s->loopLength = 2;
return true;
}