// Allocate a new sound chunk and pitch-shift an existing sound up-or-down
// into it.
static allocated_sound_t *PitchShift(allocated_sound_t *insnd, int pitch)
{
allocated_sound_t *outsnd;
Sint16 *inp, *outp;
Sint16 *srcbuf, *dstbuf;
Uint32 srclen, dstlen;
srcbuf = (Sint16 *)insnd->chunk.abuf;
srclen = insnd->chunk.alen;
// determine ratio pitch:NORM_PITCH and apply to srclen, then invert.
// This is an approximation of vanilla behaviour based on measurements
dstlen = (int)((1 + (1 - (float)pitch / NORM_PITCH)) * srclen);
// ensure that the new buffer is an even length
if (!(dstlen % 2))
++dstlen;
outsnd = AllocateSound(insnd->sfxinfo, dstlen);
if (!outsnd)
return NULL;
outsnd->pitch = pitch;
dstbuf = (Sint16 *)outsnd->chunk.abuf;
// loop over output buffer. find corresponding input cell, copy over
for (outp = dstbuf; outp < dstbuf + dstlen / 2; ++outp)
{
inp = srcbuf + (int)((float)(outp - dstbuf) / dstlen * srclen);
*outp = *inp;
}
return outsnd;
}
void SndCopyToPool()
{
for (uint i = 0; i < ORIGINAL_SAMPLE_COUNT; i++) {
SoundEntry *sound = AllocateSound();
*sound = _original_sounds[_sound_idx[i]];
sound->volume = _sound_base_vol[i];
sound->priority = 0;
}
}
static short StartQL(void)
{ short e;
e=QL_memory();
if(e==0)
{ e=LoadRoms();
if(e==0)
{ InitSerial();
RestartQL();
e=AllocateDisk();
if(e==0) e=AllocateSound();
if(e==0)
{ qlRunning=true;
StartTimer();
}
else DisposePtr((Ptr)theROM);
}
else DisposePtr((Ptr)theROM);
}
return e;
}
// Generic sound expansion function for any sample rate.
// Returns number of clipped samples (always 0).
static dboolean ExpandSoundData(sfxinfo_t *sfxinfo, byte *data, int samplerate, int length)
{
SDL_AudioCVT convertor;
allocated_sound_t *snd;
Mix_Chunk *chunk;
uint32_t expanded_length;
// Calculate the length of the expanded version of the sample.
expanded_length = (uint32_t)(((uint64_t)length * mixer_freq) / samplerate);
// Double up twice: 8 -> 16 bit and mono -> stereo
expanded_length *= 4;
// Allocate a chunk in which to expand the sound
snd = AllocateSound(sfxinfo, expanded_length);
if (!snd)
return false;
chunk = &snd->chunk;
// If we can, use the standard / optimized SDL conversion routines.
if (samplerate <= mixer_freq && ConvertibleRatio(samplerate, mixer_freq)
&& SDL_BuildAudioCVT(&convertor, AUDIO_U8, 1, samplerate, mixer_format, mixer_channels,
mixer_freq))
{
convertor.buf = chunk->abuf;
convertor.len = length;
memcpy(convertor.buf, data, length);
SDL_ConvertAudio(&convertor);
}
else
{
Sint16 *expanded = (Sint16 *)chunk->abuf;
int expand_ratio;
unsigned int i;
// Generic expansion if conversion does not work:
//
// SDL's audio conversion only works for rate conversions that are
// powers of 2; if the two formats are not in a direct power of 2
// ratio, do this naive conversion instead.
// number of samples in the converted sound
expanded_length = ((uint64_t)length * mixer_freq) / samplerate;
expand_ratio = (length << 8) / expanded_length;
for (i = 0; i < expanded_length; ++i)
{
Sint16 sample;
int src;
src = (i * expand_ratio) >> 8;
sample = (data[src] | (data[src] << 8)) - 32768;
// expand 8->16 bits, mono->stereo
expanded[i * 2] = expanded[i * 2 + 1] = sample;
}
{
float rc, dt, alpha;
// Low-pass filter for cutoff frequency f:
//
// For sampling rate r, dt = 1 / r
// rc = 1 / 2*pi*f
// alpha = dt / (rc + dt)
// Filter to the half sample rate of the original sound effect
// (maximum frequency, by nyquist)
dt = 1.0f / mixer_freq;
rc = 1.0f / (float)(2 * M_PI * samplerate);
alpha = dt / (rc + dt);
// Both channels are processed in parallel, hence [i - 2]:
for (i = 2; i < expanded_length * 2; ++i)
expanded[i] = (Sint16)(alpha * expanded[i] + (1 - alpha) * expanded[i - 2]);
}
}
return true;
}
static boolean ExpandSoundData_SRC(sfxinfo_t *sfxinfo,
byte *data,
int samplerate,
int length)
{
SRC_DATA src_data;
uint32_t i, abuf_index=0, clipped=0;
uint32_t alen;
int retn;
int16_t *expanded;
Mix_Chunk *chunk;
src_data.input_frames = length;
src_data.data_in = malloc(length * sizeof(float));
src_data.src_ratio = (double)mixer_freq / samplerate;
// We include some extra space here in case of rounding-up.
src_data.output_frames = src_data.src_ratio * length + (mixer_freq / 4);
src_data.data_out = malloc(src_data.output_frames * sizeof(float));
assert(src_data.data_in != NULL && src_data.data_out != NULL);
// Convert input data to floats
for (i=0; i<length; ++i)
{
// Unclear whether 128 should be interpreted as "zero" or whether a
// symmetrical range should be assumed. The following assumes a
// symmetrical range.
src_data.data_in[i] = data[i] / 127.5 - 1;
}
// Do the sound conversion
retn = src_simple(&src_data, SRC_ConversionMode(), 1);
assert(retn == 0);
// Allocate the new chunk.
alen = src_data.output_frames_gen * 4;
chunk = AllocateSound(sfxinfo, src_data.output_frames_gen * 4);
if (chunk == NULL)
{
return false;
}
expanded = (int16_t *) chunk->abuf;
// Convert the result back into 16-bit integers.
for (i=0; i<src_data.output_frames_gen; ++i)
{
// libsamplerate does not limit itself to the -1.0 .. 1.0 range on
// output, so a multiplier less than INT16_MAX (32767) is required
// to avoid overflows or clipping. However, the smaller the
// multiplier, the quieter the sound effects get, and the more you
// have to turn down the music to keep it in balance.
// 22265 is the largest multiplier that can be used to resample all
// of the Vanilla DOOM sound effects to 48 kHz without clipping
// using SRC_SINC_BEST_QUALITY. It is close enough (only slightly
// too conservative) for SRC_SINC_MEDIUM_QUALITY and
// SRC_SINC_FASTEST. PWADs with interestingly different sound
// effects or target rates other than 48 kHz might still result in
// clipping--I don't know if there's a limit to it.
// As the number of clipped samples increases, the signal is
// gradually overtaken by noise, with the loudest parts going first.
// However, a moderate amount of clipping is often tolerated in the
// quest for the loudest possible sound overall. The results of
// using INT16_MAX as the multiplier are not all that bad, but
// artifacts are noticeable during the loudest parts.
float cvtval_f =
src_data.data_out[i] * libsamplerate_scale * INT16_MAX;
int32_t cvtval_i = cvtval_f + (cvtval_f < 0 ? -0.5 : 0.5);
// Asymmetrical sound worries me, so we won't use -32768.
if (cvtval_i < -INT16_MAX)
{
cvtval_i = -INT16_MAX;
++clipped;
}
else if (cvtval_i > INT16_MAX)
{
cvtval_i = INT16_MAX;
++clipped;
}
// Left and right channels
expanded[abuf_index++] = cvtval_i;
expanded[abuf_index++] = cvtval_i;
}
free(src_data.data_in);
free(src_data.data_out);
if (clipped > 0)
{
fprintf(stderr, "Sound '%s': clipped %u samples (%0.2f %%)\n",
sfxinfo->name, clipped,
400.0 * clipped / chunk->alen);
}
return true;
}
