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audio: First shot at the SDL3 audio subsystem redesign!

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This is a work in progress! (and this commit will probably get
force-pushed over at some point).
This commit is contained in:
Ryan C. Gordon
2023-05-12 23:37:02 -04:00
parent b221b59995
commit 905c4fff5b
19 changed files with 1974 additions and 2464 deletions
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66
src/audio/SDL_wave.c
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@@ -1241,7 +1241,7 @@ static int LAW_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
dst = (Sint16 *)src;
/* Work backwards, since we're expanding in-place. SDL_AudioSpec.format will
/* Work backwards, since we're expanding in-place. `format` will
* inform the caller about the byte order.
*/
i = sample_count;
@@ -1667,15 +1667,13 @@ static int WaveCheckFormat(WaveFile *file, size_t datalength)
if (format->channels == 0) {
return SDL_SetError("Invalid number of channels");
} else if (format->channels > 255) {
/* Limit given by SDL_AudioSpec.channels. */
return SDL_SetError("Number of channels exceeds limit of 255");
} else if (format->channels > INT_MAX) {
return SDL_SetError("Number of channels exceeds limit of %d", INT_MAX);
}
if (format->frequency == 0) {
return SDL_SetError("Invalid sample rate");
} else if (format->frequency > INT_MAX) {
/* Limit given by SDL_AudioSpec.freq. */
return SDL_SetError("Sample rate exceeds limit of %d", INT_MAX);
}
@@ -1766,7 +1764,7 @@ static int WaveCheckFormat(WaveFile *file, size_t datalength)
return 0;
}
static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioFormat *fmt, int *channels, int *freq, Uint8 **audio_buf, Uint32 *audio_len)
{
int result;
Uint32 chunkcount = 0;
@@ -2025,13 +2023,11 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 *
break;
}
/* Setting up the SDL_AudioSpec. All unsupported formats were filtered out
/* Setting up the specs. All unsupported formats were filtered out
* by checks earlier in this function.
*/
SDL_zerop(spec);
spec->freq = format->frequency;
spec->channels = (Uint8)format->channels;
spec->samples = 4096; /* Good default buffer size */
*freq = format->frequency;
*channels = (Uint8)format->channels;
switch (format->encoding) {
case MS_ADPCM_CODE:
@@ -2039,22 +2035,22 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 *
case ALAW_CODE:
case MULAW_CODE:
/* These can be easily stored in the byte order of the system. */
spec->format = SDL_AUDIO_S16SYS;
*fmt = SDL_AUDIO_S16SYS;
break;
case IEEE_FLOAT_CODE:
spec->format = SDL_AUDIO_F32LSB;
*fmt = SDL_AUDIO_F32LSB;
break;
case PCM_CODE:
switch (format->bitspersample) {
case 8:
spec->format = SDL_AUDIO_U8;
*fmt = SDL_AUDIO_U8;
break;
case 16:
spec->format = SDL_AUDIO_S16LSB;
*fmt = SDL_AUDIO_S16LSB;
break;
case 24: /* Has been shifted to 32 bits. */
case 32:
spec->format = SDL_AUDIO_S32LSB;
*fmt = SDL_AUDIO_S32LSB;
break;
default:
/* Just in case something unexpected happened in the checks. */
@@ -2063,8 +2059,6 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 *
break;
}
spec->silence = SDL_GetSilenceValueForFormat(spec->format);
/* Report the end position back to the cleanup code. */
if (RIFFlengthknown) {
chunk->position = RIFFend;
@@ -2075,39 +2069,37 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 *
return 0;
}
SDL_AudioSpec *SDL_LoadWAV_RW(SDL_RWops *src, SDL_bool freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
int SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioFormat *fmt, int *channels, int *freq, Uint8 **audio_buf, Uint32 *audio_len)
{
int result = -1;
WaveFile file;
SDL_zero(file);
/* Make sure we are passed a valid data source */
if (src == NULL) {
/* Error may come from RWops. */
goto done;
} else if (spec == NULL) {
SDL_InvalidParamError("spec");
goto done;
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 (audio_buf == NULL) {
SDL_InvalidParamError("audio_buf");
goto done;
return SDL_InvalidParamError("audio_buf");
} else if (audio_len == NULL) {
SDL_InvalidParamError("audio_len");
goto done;
return SDL_InvalidParamError("audio_len");
}
*audio_buf = NULL;
*audio_len = 0;
SDL_zero(file);
file.riffhint = WaveGetRiffSizeHint();
file.trunchint = WaveGetTruncationHint();
file.facthint = WaveGetFactChunkHint();
result = WaveLoad(src, &file, spec, audio_buf, audio_len);
result = WaveLoad(src, &file, fmt, channels, freq, audio_buf, audio_len);
if (result < 0) {
SDL_free(*audio_buf);
spec = NULL;
audio_buf = NULL;
audio_len = 0;
}
@@ -2119,13 +2111,5 @@ SDL_AudioSpec *SDL_LoadWAV_RW(SDL_RWops *src, SDL_bool freesrc, SDL_AudioSpec *s
WaveFreeChunkData(&file.chunk);
SDL_free(file.decoderdata);
done:
if (freesrc && src) {
SDL_RWclose(src);
}
if (result == 0) {
return spec;
} else {
return NULL;
}
return result;
}