static void SDLAudioCallback(void * /*userdata*/, Uint8 *stream, int len)
{
if (playing_sample)
{
if (cursnd.sfxr)
{
uint l = len/2;
float *fbuf = new float[l];
memset(fbuf, 0, sizeof(float)*l);
SynthSample(l, fbuf, nullptr);
while (l--)
{
float f = fbuf[l];
if (f < -1.0) f = -1.0;
if (f > 1.0) f = 1.0;
((Sint16*)stream)[l] = (Sint16)(f * 32767);
}
delete[] fbuf;
}
else
{
size_t amount = min((size_t)len, cursnd.len - (cursndpos - cursnd.buf));
memcpy(stream, cursndpos, amount);
memset(stream + amount, 0, len - amount);
cursndpos += amount;
if (cursndpos == cursnd.buf + cursnd.len)
playing_sample = false;
}
}
else
{
memset(stream, 0, len);
SDL_PauseAudioDevice(audioid, 1);
}
}
bool ExportWAV(char* filename)
{
FILE* foutput=fopen(filename, "wb");
if(!foutput)
return false;
// write wav header
char string[32];
unsigned int dword=0;
unsigned short word=0;
fwrite("RIFF", 4, 1, foutput); // "RIFF"
dword=0;
fwrite(&dword, 1, 4, foutput); // remaining file size
fwrite("WAVE", 4, 1, foutput); // "WAVE"
fwrite("fmt ", 4, 1, foutput); // "fmt "
dword=16;
fwrite(&dword, 1, 4, foutput); // chunk size
word=1;
fwrite(&word, 1, 2, foutput); // compression code
word=1;
fwrite(&word, 1, 2, foutput); // channels
dword=wav_freq;
fwrite(&dword, 1, 4, foutput); // sample rate
dword=wav_freq*wav_bits/8;
fwrite(&dword, 1, 4, foutput); // bytes/sec
word=wav_bits/8;
fwrite(&word, 1, 2, foutput); // block align
word=wav_bits;
fwrite(&word, 1, 2, foutput); // bits per sample
fwrite("data", 4, 1, foutput); // "data"
dword=0;
int foutstream_datasize=ftell(foutput);
fwrite(&dword, 1, 4, foutput); // chunk size
// write sample data
mute_stream=true;
file_sampleswritten=0;
filesample=0.0f;
fileacc=0;
PlaySample();
while(playing_sample)
SynthSample(256, NULL, foutput);
mute_stream=false;
// seek back to header and write size info
fseek(foutput, 4, SEEK_SET);
dword=0;
dword=foutstream_datasize-4+file_sampleswritten*wav_bits/8;
fwrite(&dword, 1, 4, foutput); // remaining file size
fseek(foutput, foutstream_datasize, SEEK_SET);
dword=file_sampleswritten*wav_bits/8;
fwrite(&dword, 1, 4, foutput); // chunk size (data)
fclose(foutput);
return true;
}
//ancient portaudio stuff
static int AudioCallback(void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData)
{
float *out=(float*)outputBuffer;
float *in=(float*)inputBuffer;
(void)outTime;
if(playing_sample && !mute_stream)
SynthSample(framesPerBuffer, out, NULL);
else
for(int i=0;i<framesPerBuffer;i++)
*out++=0.0f;
return 0;
}
//lets use SDL in stead
static void SDLAudioCallback(void *userdata, Uint8 *stream, int len)
{
if (playing_sample && !mute_stream)
{
unsigned int l = len/2;
float fbuf[l];
memset(fbuf, 0, sizeof(fbuf));
SynthSample(l, fbuf, NULL);
while (l--)
{
float f = fbuf[l];
if (f < -1.0) f = -1.0;
if (f > 1.0) f = 1.0;
((Sint16*)stream)[l] = (Sint16)(f * 32767);
}
}
else memset(stream, 0, len);
}
// This is a replacement for the original sfxr callback function.
bool sfxr::operator ()(unsigned char* sampleBuffer, int byteCount)
{
if (playing_sample && !mute_stream)
{
unsigned int l = byteCount/2;
float fbuf[l];
memset(fbuf, 0, sizeof(fbuf));
SynthSample(l, fbuf, NULL);
while (l--)
{
float f = fbuf[l];
if (f < -1.0) f = -1.0;
if (f > 1.0) f = 1.0;
((sint16*)sampleBuffer)[l] = (sint16)(f * 32767);
}
return true;
}
memset(sampleBuffer, 0, byteCount);
return false;
}
//lets use SDL in stead
void generator::GenerateAudio(int16_t * bufferInterlace, int32_t nbSample, int32_t nbChannels)
{
if (playing_sample) {
float fbuf[nbSample];
memset(fbuf, 0, sizeof(float));
SynthSample(nbSample, fbuf, NULL);
while (nbSample--) {
float f = fbuf[nbSample];
if (f < -1.0) {
f = -1.0;
}
if (f > 1.0) {
f = 1.0;
}
for (int32_t iii=0; iii<nbChannels; iii++) {
bufferInterlace[nbSample*nbChannels+iii] = (int16_t)(f * 32767);
}
}
}
}
