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Update for SDL3 coding style (#6717)

Browse Source 
I updated .clang-format and ran clang-format 14 over the src and test directories to standardize the code base.

In general I let clang-format have it's way, and added markup to prevent formatting of code that would break or be completely unreadable if formatted.

The script I ran for the src directory is added as build-scripts/clang-format-src.sh

This fixes:
#6592
#6593
#6594
This commit is contained in:
Sam Lantinga
2022-11-30 12:51:59 -08:00
committed by GitHub
parent 14b902faca
commit 5750bcb174
781 changed files with 51659 additions and 55763 deletions
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406
src/audio/SDL_mixer.c
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@@ -78,14 +78,12 @@ static const Uint8 mix8[] = {
};
/* The volume ranges from 0 - 128 */
#define ADJUST_VOLUME(s, v) (s = (s*v)/SDL_MIX_MAXVOLUME)
#define ADJUST_VOLUME_U8(s, v) (s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128)
#define ADJUST_VOLUME_U16(s, v) (s = (((s-32768)*v)/SDL_MIX_MAXVOLUME)+32768)
#define ADJUST_VOLUME(s, v) (s = (s * v) / SDL_MIX_MAXVOLUME)
#define ADJUST_VOLUME_U8(s, v) (s = (((s - 128) * v) / SDL_MIX_MAXVOLUME) + 128)
#define ADJUST_VOLUME_U16(s, v) (s = (((s - 32768) * v) / SDL_MIX_MAXVOLUME) + 32768)
void
SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
Uint32 len, int volume)
void SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,
Uint32 len, int volume)
{
if (volume == 0) {
return;
@@ -94,258 +92,248 @@ SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
switch (format) {
case AUDIO_U8:
{
Uint8 src_sample;
{
Uint8 src_sample;
while (len--) {
src_sample = *src;
ADJUST_VOLUME_U8(src_sample, volume);
*dst = mix8[*dst + src_sample];
++dst;
++src;
}
while (len--) {
src_sample = *src;
ADJUST_VOLUME_U8(src_sample, volume);
*dst = mix8[*dst + src_sample];
++dst;
++src;
}
break;
} break;
case AUDIO_S8:
{
Sint8 *dst8, *src8;
Sint8 src_sample;
int dst_sample;
const int max_audioval = SDL_MAX_SINT8;
const int min_audioval = SDL_MIN_SINT8;
{
Sint8 *dst8, *src8;
Sint8 src_sample;
int dst_sample;
const int max_audioval = SDL_MAX_SINT8;
const int min_audioval = SDL_MIN_SINT8;
src8 = (Sint8 *) src;
dst8 = (Sint8 *) dst;
while (len--) {
src_sample = *src8;
ADJUST_VOLUME(src_sample, volume);
dst_sample = *dst8 + src_sample;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*dst8 = dst_sample;
++dst8;
++src8;
src8 = (Sint8 *)src;
dst8 = (Sint8 *)dst;
while (len--) {
src_sample = *src8;
ADJUST_VOLUME(src_sample, volume);
dst_sample = *dst8 + src_sample;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*dst8 = dst_sample;
++dst8;
++src8;
}
break;
} break;
case AUDIO_S16LSB:
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapLE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapLE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_S16MSB:
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Sint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapBE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Sint16 *)src);
ADJUST_VOLUME(src1, volume);
src2 = SDL_SwapBE16(*(Sint16 *)dst);
src += 2;
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(Sint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_U16LSB:
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapLE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapLE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapLE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapLE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_U16MSB:
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
{
Uint16 src1, src2;
int dst_sample;
const int max_audioval = SDL_MAX_SINT16;
const int min_audioval = SDL_MIN_SINT16;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapBE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
len /= 2;
while (len--) {
src1 = SDL_SwapBE16(*(Uint16 *)src);
ADJUST_VOLUME_U16(src1, volume);
src2 = SDL_SwapBE16(*(Uint16 *)dst);
src += 2;
dst_sample = src1 + src2 - 32768 * 2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
dst_sample += 32768;
*(Uint16 *)dst = SDL_SwapBE16(dst_sample);
dst += 2;
}
break;
} break;
case AUDIO_S32LSB:
{
const Uint32 *src32 = (Uint32 *) src;
Uint32 *dst32 = (Uint32 *) dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
{
const Uint32 *src32 = (Uint32 *)src;
Uint32 *dst32 = (Uint32 *)dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
len /= 4;
while (len--) {
src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
len /= 4;
while (len--) {
src1 = (Sint64)((Sint32)SDL_SwapLE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64)((Sint32)SDL_SwapLE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapLE32((Uint32)((Sint32)dst_sample));
}
break;
} break;
case AUDIO_S32MSB:
{
const Uint32 *src32 = (Uint32 *) src;
Uint32 *dst32 = (Uint32 *) dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
{
const Uint32 *src32 = (Uint32 *)src;
Uint32 *dst32 = (Uint32 *)dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = SDL_MAX_SINT32;
const Sint64 min_audioval = SDL_MIN_SINT32;
len /= 4;
while (len--) {
src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
len /= 4;
while (len--) {
src1 = (Sint64)((Sint32)SDL_SwapBE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64)((Sint32)SDL_SwapBE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapBE32((Uint32)((Sint32)dst_sample));
}
break;
} break;
case AUDIO_F32LSB:
{
const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
const float fvolume = (float) volume;
const float *src32 = (float *) src;
float *dst32 = (float *) dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
{
const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME);
const float fvolume = (float)volume;
const float *src32 = (float *)src;
float *dst32 = (float *)dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatLE(*dst32);
src32++;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatLE(*dst32);
src32++;
dst_sample = ((double) src1) + ((double) src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatLE((float) dst_sample);
dst_sample = ((double)src1) + ((double)src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatLE((float)dst_sample);
}
break;
} break;
case AUDIO_F32MSB:
{
const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
const float fvolume = (float) volume;
const float *src32 = (float *) src;
float *dst32 = (float *) dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
{
const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME);
const float fvolume = (float)volume;
const float *src32 = (float *)src;
float *dst32 = (float *)dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatBE(*dst32);
src32++;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatBE(*dst32);
src32++;
dst_sample = ((double) src1) + ((double) src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatBE((float) dst_sample);
dst_sample = ((double)src1) + ((double)src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatBE((float)dst_sample);
}
break;
} break;
default: /* If this happens... FIXME! */
default: /* If this happens... FIXME! */
SDL_SetError("SDL_MixAudioFormat(): unknown audio format");
return;
}