static boolean ConvertibleRatio(int freq1, int freq2)
{
int ratio;
if (freq1 > freq2)
{
return ConvertibleRatio(freq2, freq1);
}
else if ((freq2 % freq1) != 0)
{
// Not in a direct ratio
return false;
}
else
{
// Check the ratio is a power of 2
ratio = freq2 / freq1;
while ((ratio & 1) == 0)
{
ratio = ratio >> 1;
}
return ratio == 1;
}
}
static void ExpandSoundData(byte *data,
int samplerate,
int length,
Mix_Chunk *destination)
{
SDL_AudioCVT convertor;
if (samplerate <= mixer_freq
&& ConvertibleRatio(samplerate, mixer_freq)
&& SDL_BuildAudioCVT(&convertor,
AUDIO_U8, 1, samplerate,
mixer_format, mixer_channels, mixer_freq))
{
convertor.buf = destination->abuf;
convertor.len = length;
memcpy(convertor.buf, data, length);
SDL_ConvertAudio(&convertor);
}
else
{
Sint16 *expanded = (Sint16 *) destination->abuf;
int expanded_length;
int expand_ratio;
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);
sample -= 32768;
// expand 8->16 bits, mono->stereo
expanded[i * 2] = expanded[i * 2 + 1] = sample;
}
}
}
static dboolean ConvertibleRatio(int freq1, int freq2)
{
int ratio;
if (freq1 > freq2)
return ConvertibleRatio(freq2, freq1);
else if (freq2 % freq1)
return false; // Not in a direct ratio
else
{
// Check the ratio is a power of 2
ratio = freq2 / freq1;
while (!(ratio & 1))
ratio >>= 1;
return (ratio == 1);
}
}
// Generic sound expansion function for any sample rate.
static void ExpandSoundData_SDL(byte *data, int samplerate,
uint32_t length, Mix_Chunk *destination)
{
SDL_AudioCVT convertor;
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;
destination->alen = expanded_length;
destination->abuf = Z_Malloc(expanded_length, PU_STATIC, (void **)&destination->abuf);
// 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 = destination->abuf;
convertor.len = length;
memcpy(convertor.buf, data, length);
SDL_ConvertAudio(&convertor);
}
else
{
Sint16 *expanded = (Sint16 *)destination->abuf;
int expanded_length;
int expand_ratio;
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]);
}
}
}
static boolean ExpandSoundData_SDL(sfxinfo_t *sfxinfo,
byte *data,
int samplerate,
int length)
{
SDL_AudioCVT convertor;
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
chunk = AllocateSound(sfxinfo, expanded_length);
if (chunk == NULL)
{
return false;
}
// 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 expanded_length;
int expand_ratio;
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);
sample -= 32768;
// expand 8->16 bits, mono->stereo
expanded[i * 2] = expanded[i * 2 + 1] = sample;
}
#ifdef LOW_PASS_FILTER
// Perform a low-pass filter on the upscaled sound to filter
// out high-frequency noise from the conversion process.
{
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 / (3.14f * 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]);
}
}
#endif /* #ifdef LOW_PASS_FILTER */
}
return true;
}
