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audio: Readd SDL_AudioSpec, but just with format/channels/freq fields.

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This commit is contained in:
Ryan C. Gordon
2023-05-30 01:08:22 -04:00
parent e6aaed7d79
commit 26525f5fd3
12 changed files with 244 additions and 287 deletions
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29
src/audio/SDL_wave.c
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@@ -1764,7 +1764,7 @@ static int WaveCheckFormat(WaveFile *file, size_t datalength)
return 0;
}
static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioFormat *fmt, int *channels, int *freq, Uint8 **audio_buf, Uint32 *audio_len)
static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
{
int result;
Uint32 chunkcount = 0;
@@ -2026,8 +2026,9 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioFormat *fmt, int *c
/* Setting up the specs. All unsupported formats were filtered out
* by checks earlier in this function.
*/
*freq = format->frequency;
*channels = (Uint8)format->channels;
spec->freq = format->frequency;
spec->channels = (Uint8)format->channels;
spec->format = 0;
switch (format->encoding) {
case MS_ADPCM_CODE:
@@ -2035,22 +2036,22 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioFormat *fmt, int *c
case ALAW_CODE:
case MULAW_CODE:
/* These can be easily stored in the byte order of the system. */
*fmt = SDL_AUDIO_S16SYS;
spec->format = SDL_AUDIO_S16SYS;
break;
case IEEE_FLOAT_CODE:
*fmt = SDL_AUDIO_F32LSB;
spec->format = SDL_AUDIO_F32LSB;
break;
case PCM_CODE:
switch (format->bitspersample) {
case 8:
*fmt = SDL_AUDIO_U8;
spec->format = SDL_AUDIO_U8;
break;
case 16:
*fmt = SDL_AUDIO_S16LSB;
spec->format = SDL_AUDIO_S16LSB;
break;
case 24: /* Has been shifted to 32 bits. */
case 32:
*fmt = SDL_AUDIO_S32LSB;
spec->format = SDL_AUDIO_S32LSB;
break;
default:
/* Just in case something unexpected happened in the checks. */
@@ -2069,7 +2070,7 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioFormat *fmt, int *c
return 0;
}
int SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioFormat *fmt, int *channels, int *freq, Uint8 **audio_buf, Uint32 *audio_len)
int SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
{
int result = -1;
WaveFile file;
@@ -2077,12 +2078,8 @@ int SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioFormat *fmt, int *chann
/* Make sure we are passed a valid data source */
if (src == NULL) {
return -1; /* Error may come from RWops. */
} else if (fmt == NULL) {
return SDL_InvalidParamError("fmt");
} else if (channels == NULL) {
return SDL_InvalidParamError("channels");
} else if (freq == NULL) {
return SDL_InvalidParamError("freq");
} else if (spec == NULL) {
return SDL_InvalidParamError("spec");
} else if (audio_buf == NULL) {
return SDL_InvalidParamError("audio_buf");
} else if (audio_len == NULL) {
@@ -2097,7 +2094,7 @@ int SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioFormat *fmt, int *chann
file.trunchint = WaveGetTruncationHint();
file.facthint = WaveGetFactChunkHint();
result = WaveLoad(src, &file, fmt, channels, freq, audio_buf, audio_len);
result = WaveLoad(src, &file, spec, audio_buf, audio_len);
if (result < 0) {
SDL_free(*audio_buf);
audio_buf = NULL;