Mercurial > sdl-ios-xcode
diff src/audio/SDL_mixer.c @ 1982:3b4ce57c6215
First shot at new audio data types (int32 and float32).
Notable changes:
- Converters between types are autogenerated. Instead of making multiple
passes over the data with seperate filters for endianess, size, signedness,
etc, converting between data types is always one specialized filter. This
simplifies SDL_BuildAudioCVT(), which otherwise had a million edge cases
with the new types, and makes the actually conversions more CPU cache
friendly. Left a stub for adding specific optimized versions of these
routines (SSE/MMX/Altivec, assembler, etc)
- Autogenerated converters are built by SDL/src/audio/sdlgenaudiocvt.pl. This
does not need to be run unless tweaking the code, and thus doesn't need
integration into the build system.
- Went through all the drivers and tried to weed out all the "Uint16"
references that are better specified with the new SDL_AudioFormat typedef.
- Cleaned out a bunch of hardcoded bitwise magic numbers and replaced them
with new SDL_AUDIO_* macros.
- Added initial float32 and int32 support code. Theoretically, existing
drivers will push these through converters to get the data they want to
feed to the hardware.
Still TODO:
- Optimize and debug new converters.
- Update the CoreAudio backend to accept float32 data directly.
- Other backends, too?
- SDL_LoadWAV() needs to be updated to support int32 and float32 .wav files
(both of which exist and can be generated by 'sox' for testing purposes).
- Update the mixer to handle new datatypes.
- Optionally update SDL_sound and SDL_mixer, etc.
| author | Ryan C. Gordon <icculus@icculus.org> |
|---|---|
| date | Thu, 24 Aug 2006 12:10:46 +0000 |
| parents | c121d94672cb |
| children | 8055185ae4ed |
line wrap: on
line diff
--- a/src/audio/SDL_mixer.c Thu Aug 10 15:15:06 2006 +0000 +++ b/src/audio/SDL_mixer.c Thu Aug 24 12:10:46 2006 +0000 @@ -92,22 +92,27 @@ void SDL_MixAudio(Uint8 * dst, const Uint8 * src, Uint32 len, int volume) { - Uint16 format; - - if (volume == 0) { - return; - } /* Mix the user-level audio format */ if (current_audio) { + SDL_AudioFormat format; if (current_audio->convert.needed) { format = current_audio->convert.src_format; } else { format = current_audio->spec.format; } - } else { - /* HACK HACK HACK */ - format = AUDIO_S16; + SDL_MixAudioFormat(dst, src, format, len, volume); } +} + + +void +SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format, + Uint32 len, int volume) +{ + if (volume == 0) { + return; + } + switch (format) { case AUDIO_U8: @@ -252,6 +257,134 @@ } break; + case AUDIO_S32LSB: + { + const Uint32 *src32 = (Uint32 *) src; + Uint32 *dst32 = (Uint32 *) dst; + Sint32 src1, src2; + Sint64 dst_sample; + const Sint64 max_audioval = ((((Sint64)1) << (32 - 1)) - 1); + const Sint64 min_audioval = -(((Sint64)1) << (32 - 1)); + + len /= 4; + while (len--) { + src1 = (Sint32) SDL_SwapLE32(*src32); + src32++; + ADJUST_VOLUME(src1, volume); + src2 = (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; + + case AUDIO_S32MSB: + { + const Uint32 *src32 = (Uint32 *) src; + Uint32 *dst32 = (Uint32 *) dst; + Sint32 src1, src2; + Sint64 dst_sample; + const Sint64 max_audioval = ((((Sint64)1) << (32 - 1)) - 1); + const Sint64 min_audioval = -(((Sint64)1) << (32 - 1)); + + len /= 4; + while (len--) { + src1 = (Sint32) SDL_SwapBE32(*src32); + src32++; + ADJUST_VOLUME(src1, volume); + src2 = (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; + + 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 = 3.40282347e+38F; + const double min_audioval = -3.40282347e+38F; + + /* !!! FIXME: this is a little nasty. */ + union { float f; Uint32 ui32; } cvt; + + len /= 4; + while (len--) { + cvt.f = *(src32++); + cvt.ui32 = SDL_SwapLE32(cvt.ui32); + src1 = ((cvt.f * fvolume) * fmaxvolume); + + cvt.f = *dst32; + cvt.ui32 = SDL_SwapLE32(cvt.ui32); + src2 = cvt.f; + + dst_sample = src1 + src2; + if (dst_sample > max_audioval) { + dst_sample = max_audioval; + } else if (dst_sample < min_audioval) { + dst_sample = min_audioval; + } + cvt.f = ((float) dst_sample); + cvt.ui32 = SDL_SwapLE32(cvt.ui32); + *(dst32++) = cvt.f; + } + } + 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 = 3.40282347e+38F; + const double min_audioval = -3.40282347e+38F; + + /* !!! FIXME: this is a little nasty. */ + union { float f; Uint32 ui32; } cvt; + + len /= 4; + while (len--) { + cvt.f = *(src32++); + cvt.ui32 = SDL_SwapBE32(cvt.ui32); + src1 = ((cvt.f * fvolume) * fmaxvolume); + + cvt.f = *dst32; + cvt.ui32 = SDL_SwapBE32(cvt.ui32); + src2 = cvt.f; + + dst_sample = src1 + src2; + if (dst_sample > max_audioval) { + dst_sample = max_audioval; + } else if (dst_sample < min_audioval) { + dst_sample = min_audioval; + } + cvt.f = ((float) dst_sample); + cvt.ui32 = SDL_SwapBE32(cvt.ui32); + *(dst32++) = cvt.f; + } + } + break; + default: /* If this happens... FIXME! */ SDL_SetError("SDL_MixAudio(): unknown audio format"); return;