Mercurial > sdl-ios-xcode
diff src/audio/SDL_audiocvt.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_audiocvt.c Thu Aug 10 15:15:06 2006 +0000 +++ b/src/audio/SDL_audiocvt.c Thu Aug 24 12:10:46 2006 +0000 @@ -24,11 +24,11 @@ /* Functions for audio drivers to perform runtime conversion of audio format */ #include "SDL_audio.h" - +#include "SDL_audio_c.h" /* Effectively mix right and left channels into a single channel */ -void SDLCALL -SDL_ConvertMono(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; Sint32 sample; @@ -36,8 +36,7 @@ #ifdef DEBUG_CONVERT fprintf(stderr, "Converting to mono\n"); #endif - switch (format & 0x8018) { - + switch (format & (SDL_AUDIO_MASK_SIGNED|SDL_AUDIO_MASK_BITSIZE)) { case AUDIO_U8: { Uint8 *src, *dst; @@ -84,7 +83,7 @@ src = cvt->buf; dst = cvt->buf; - if ((format & 0x1000) == 0x1000) { + if (SDL_AUDIO_ISBIGENDIAN(format)) { for (i = cvt->len_cvt / 4; i; --i) { sample = (Uint16) ((src[0] << 8) | src[1]) + (Uint16) ((src[2] << 8) | src[3]); @@ -124,7 +123,7 @@ src = cvt->buf; dst = cvt->buf; - if ((format & 0x1000) == 0x1000) { + if (SDL_AUDIO_ISBIGENDIAN(format)) { for (i = cvt->len_cvt / 4; i; --i) { sample = (Sint16) ((src[0] << 8) | src[1]) + (Sint16) ((src[2] << 8) | src[3]); @@ -163,127 +162,106 @@ } } break; + + case AUDIO_S32: + { + const Uint32 *src = (const Uint32 *) cvt->buf; + Uint32 *dst = (Uint32 *) cvt->buf; + if (SDL_AUDIO_ISBIGENDIAN(format)) { + for (i = cvt->len_cvt / 8; i; --i, src += 2) { + const Sint64 added = + (((Sint64) (Sint32) SDL_SwapBE32(src[0])) + + ((Sint64) (Sint32) SDL_SwapBE32(src[1]))); + *(dst++) = SDL_SwapBE32((Uint32) ((Sint32) (added >> 1))); + } + } else { + for (i = cvt->len_cvt / 8; i; --i, src += 2) { + const Sint64 added = + (((Sint64) (Sint32) SDL_SwapLE32(src[0])) + + ((Sint64) (Sint32) SDL_SwapLE32(src[1]))); + *(dst++) = SDL_SwapLE32((Uint32) ((Sint32) (added >> 1))); + } + } + } + break; + + case AUDIO_F32: + { + /* !!! FIXME: this convert union is nasty. */ + union { float f; Uint32 ui32; } f2i; + const Uint32 *src = (const Uint32 *) cvt->buf; + Uint32 *dst = (Uint32 *) cvt->buf; + if (SDL_AUDIO_ISBIGENDIAN(format)) { + for (i = cvt->len_cvt / 8; i; --i, src += 2) { + float src1, src2; + f2i.ui32 = SDL_SwapBE32(src[0]); + src1 = f2i.f; + f2i.ui32 = SDL_SwapBE32(src[1]); + src2 = f2i.f; + const double added = ((double) src1) + ((double) src2); + f2i.f = (float) (added * 0.5); + *(dst++) = SDL_SwapBE32(f2i.ui32); + } + } else { + for (i = cvt->len_cvt / 8; i; --i, src += 2) { + float src1, src2; + f2i.ui32 = SDL_SwapLE32(src[0]); + src1 = f2i.f; + f2i.ui32 = SDL_SwapLE32(src[1]); + src2 = f2i.f; + const double added = ((double) src1) + ((double) src2); + f2i.f = (float) (added * 0.5); + *(dst++) = SDL_SwapLE32(f2i.ui32); + } + } + } + break; } + cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } + /* Discard top 4 channels */ -void SDLCALL -SDL_ConvertStrip(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Sint32 lsample, rsample; #ifdef DEBUG_CONVERT - fprintf(stderr, "Converting down to stereo\n"); + fprintf(stderr, "Converting down from 6 channels to stereo\n"); #endif - switch (format & 0x8018) { - - case AUDIO_U8: - { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - for (i = cvt->len_cvt / 6; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 6; - dst += 2; - } - } - break; - case AUDIO_S8: - { - Sint8 *src, *dst; - - src = (Sint8 *) cvt->buf; - dst = (Sint8 *) cvt->buf; - for (i = cvt->len_cvt / 6; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 6; - dst += 2; - } - } - break; - - case AUDIO_U16: - { - Uint8 *src, *dst; + #define strip_chans_6_to_2(type) \ + { \ + const type *src = (const type *) cvt->buf; \ + type *dst = (type *) cvt->buf; \ + for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \ + dst[0] = src[0]; \ + dst[1] = src[1]; \ + src += 6; \ + dst += 2; \ + } \ + } - src = cvt->buf; - dst = cvt->buf; - if ((format & 0x1000) == 0x1000) { - for (i = cvt->len_cvt / 12; i; --i) { - lsample = (Uint16) ((src[0] << 8) | src[1]); - rsample = (Uint16) ((src[2] << 8) | src[3]); - dst[1] = (lsample & 0xFF); - lsample >>= 8; - dst[0] = (lsample & 0xFF); - dst[3] = (rsample & 0xFF); - rsample >>= 8; - dst[2] = (rsample & 0xFF); - src += 12; - dst += 4; - } - } else { - for (i = cvt->len_cvt / 12; i; --i) { - lsample = (Uint16) ((src[1] << 8) | src[0]); - rsample = (Uint16) ((src[3] << 8) | src[2]); - dst[0] = (lsample & 0xFF); - lsample >>= 8; - dst[1] = (lsample & 0xFF); - dst[2] = (rsample & 0xFF); - rsample >>= 8; - dst[3] = (rsample & 0xFF); - src += 12; - dst += 4; - } - } - } - break; + /* this function only cares about typesize, and data as a block of bits. */ + switch (SDL_AUDIO_BITSIZE(format)) { + case 8: + strip_chans_6_to_2(Uint8); + break; + case 16: + strip_chans_6_to_2(Uint16); + break; + case 32: + strip_chans_6_to_2(Uint32); + break; + } - case AUDIO_S16: - { - Uint8 *src, *dst; + #undef strip_chans_6_to_2 - src = cvt->buf; - dst = cvt->buf; - if ((format & 0x1000) == 0x1000) { - for (i = cvt->len_cvt / 12; i; --i) { - lsample = (Sint16) ((src[0] << 8) | src[1]); - rsample = (Sint16) ((src[2] << 8) | src[3]); - dst[1] = (lsample & 0xFF); - lsample >>= 8; - dst[0] = (lsample & 0xFF); - dst[3] = (rsample & 0xFF); - rsample >>= 8; - dst[2] = (rsample & 0xFF); - src += 12; - dst += 4; - } - } else { - for (i = cvt->len_cvt / 12; i; --i) { - lsample = (Sint16) ((src[1] << 8) | src[0]); - rsample = (Sint16) ((src[3] << 8) | src[2]); - dst[0] = (lsample & 0xFF); - lsample >>= 8; - dst[1] = (lsample & 0xFF); - dst[2] = (rsample & 0xFF); - rsample >>= 8; - dst[3] = (rsample & 0xFF); - src += 12; - dst += 4; - } - } - } - break; - } cvt->len_cvt /= 3; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -292,157 +270,87 @@ /* Discard top 2 channels of 6 */ -void SDLCALL -SDL_ConvertStrip_2(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_ConvertStrip_2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Sint32 lsample, rsample; #ifdef DEBUG_CONVERT fprintf(stderr, "Converting 6 down to quad\n"); #endif - switch (format & 0x8018) { - case AUDIO_U8: - { - Uint8 *src, *dst; - - src = cvt->buf; - dst = cvt->buf; - for (i = cvt->len_cvt / 4; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } - } - break; - - case AUDIO_S8: - { - Sint8 *src, *dst; - - src = (Sint8 *) cvt->buf; - dst = (Sint8 *) cvt->buf; - for (i = cvt->len_cvt / 4; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } - } - break; - - case AUDIO_U16: - { - Uint8 *src, *dst; + #define strip_chans_6_to_4(type) \ + { \ + const type *src = (const type *) cvt->buf; \ + type *dst = (type *) cvt->buf; \ + for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \ + dst[0] = src[0]; \ + dst[1] = src[1]; \ + dst[2] = src[2]; \ + dst[3] = src[3]; \ + src += 6; \ + dst += 4; \ + } \ + } - src = cvt->buf; - dst = cvt->buf; - if ((format & 0x1000) == 0x1000) { - for (i = cvt->len_cvt / 8; i; --i) { - lsample = (Uint16) ((src[0] << 8) | src[1]); - rsample = (Uint16) ((src[2] << 8) | src[3]); - dst[1] = (lsample & 0xFF); - lsample >>= 8; - dst[0] = (lsample & 0xFF); - dst[3] = (rsample & 0xFF); - rsample >>= 8; - dst[2] = (rsample & 0xFF); - src += 8; - dst += 4; - } - } else { - for (i = cvt->len_cvt / 8; i; --i) { - lsample = (Uint16) ((src[1] << 8) | src[0]); - rsample = (Uint16) ((src[3] << 8) | src[2]); - dst[0] = (lsample & 0xFF); - lsample >>= 8; - dst[1] = (lsample & 0xFF); - dst[2] = (rsample & 0xFF); - rsample >>= 8; - dst[3] = (rsample & 0xFF); - src += 8; - dst += 4; - } - } - } - break; + /* this function only cares about typesize, and data as a block of bits. */ + switch (SDL_AUDIO_BITSIZE(format)) { + case 8: + strip_chans_6_to_4(Uint8); + break; + case 16: + strip_chans_6_to_4(Uint16); + break; + case 32: + strip_chans_6_to_4(Uint32); + break; + } - case AUDIO_S16: - { - Uint8 *src, *dst; + #undef strip_chans_6_to_4 - src = cvt->buf; - dst = cvt->buf; - if ((format & 0x1000) == 0x1000) { - for (i = cvt->len_cvt / 8; i; --i) { - lsample = (Sint16) ((src[0] << 8) | src[1]); - rsample = (Sint16) ((src[2] << 8) | src[3]); - dst[1] = (lsample & 0xFF); - lsample >>= 8; - dst[0] = (lsample & 0xFF); - dst[3] = (rsample & 0xFF); - rsample >>= 8; - dst[2] = (rsample & 0xFF); - src += 8; - dst += 4; - } - } else { - for (i = cvt->len_cvt / 8; i; --i) { - lsample = (Sint16) ((src[1] << 8) | src[0]); - rsample = (Sint16) ((src[3] << 8) | src[2]); - dst[0] = (lsample & 0xFF); - lsample >>= 8; - dst[1] = (lsample & 0xFF); - dst[2] = (rsample & 0xFF); - rsample >>= 8; - dst[3] = (rsample & 0xFF); - src += 8; - dst += 4; - } - } - } - break; - } - cvt->len_cvt /= 2; + cvt->len_cvt /= 6; + cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } /* Duplicate a mono channel to both stereo channels */ -void SDLCALL -SDL_ConvertStereo(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_ConvertStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; #ifdef DEBUG_CONVERT fprintf(stderr, "Converting to stereo\n"); #endif - if ((format & 0xFF) == 16) { - Uint16 *src, *dst; + + #define dup_chans_1_to_2(type) \ + { \ + const type *src = (const type *) (cvt->buf + cvt->len_cvt); \ + type *dst = (type *) (cvt->buf + cvt->len_cvt * 2); \ + for (i = cvt->len_cvt / 2; i; --i, --src) { \ + const type val = *src; \ + dst -= 2; \ + dst[0] = dst[1] = val; \ + } \ + } - src = (Uint16 *) (cvt->buf + cvt->len_cvt); - dst = (Uint16 *) (cvt->buf + cvt->len_cvt * 2); - for (i = cvt->len_cvt / 2; i; --i) { - dst -= 2; - src -= 1; - dst[0] = src[0]; - dst[1] = src[0]; - } - } else { - Uint8 *src, *dst; + /* this function only cares about typesize, and data as a block of bits. */ + switch (SDL_AUDIO_BITSIZE(format)) { + case 8: + dup_chans_1_to_2(Uint8); + break; + case 16: + dup_chans_1_to_2(Uint16); + break; + case 32: + dup_chans_1_to_2(Uint32); + break; + } - src = cvt->buf + cvt->len_cvt; - dst = cvt->buf + cvt->len_cvt * 2; - for (i = cvt->len_cvt; i; --i) { - dst -= 2; - src -= 1; - dst[0] = src[0]; - dst[1] = src[0]; - } - } + #undef dup_chans_1_to_2 + cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -451,16 +359,16 @@ /* Duplicate a stereo channel to a pseudo-5.1 stream */ -void SDLCALL -SDL_ConvertSurround(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_ConvertSurround(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; #ifdef DEBUG_CONVERT fprintf(stderr, "Converting stereo to surround\n"); #endif - switch (format & 0x8018) { + switch (format & (SDL_AUDIO_MASK_SIGNED|SDL_AUDIO_MASK_BITSIZE)) { case AUDIO_U8: { Uint8 *src, *dst, lf, rf, ce; @@ -513,7 +421,7 @@ src = cvt->buf + cvt->len_cvt; dst = cvt->buf + cvt->len_cvt * 3; - if ((format & 0x1000) == 0x1000) { + if (SDL_AUDIO_ISBIGENDIAN(format)) { for (i = cvt->len_cvt / 4; i; --i) { dst -= 12; src -= 4; @@ -573,7 +481,7 @@ src = cvt->buf + cvt->len_cvt; dst = cvt->buf + cvt->len_cvt * 3; - if ((format & 0x1000) == 0x1000) { + if (SDL_AUDIO_ISBIGENDIAN(format)) { for (i = cvt->len_cvt / 4; i; --i) { dst -= 12; src -= 4; @@ -624,6 +532,96 @@ } } break; + + case AUDIO_S32: + { + Sint32 lf, rf, ce; + const Uint32 *src = (const Uint32 *) cvt->buf + cvt->len_cvt; + Uint32 *dst = (Uint32 *) cvt->buf + cvt->len_cvt * 3; + + if (SDL_AUDIO_ISBIGENDIAN(format)) { + for (i = cvt->len_cvt / 8; i; --i) { + dst -= 6; + src -= 2; + lf = (Sint32) SDL_SwapBE32(src[0]); + rf = (Sint32) SDL_SwapBE32(src[1]); + ce = (lf / 2) + (rf / 2); + dst[0] = SDL_SwapBE32((Uint32) lf); + dst[1] = SDL_SwapBE32((Uint32) rf); + dst[2] = SDL_SwapBE32((Uint32) (lf - ce)); + dst[3] = SDL_SwapBE32((Uint32) (rf - ce)); + dst[4] = SDL_SwapBE32((Uint32) ce); + dst[5] = SDL_SwapBE32((Uint32) ce); + } + } else { + for (i = cvt->len_cvt / 8; i; --i) { + dst -= 6; + src -= 2; + lf = (Sint32) SDL_SwapLE32(src[0]); + rf = (Sint32) SDL_SwapLE32(src[1]); + ce = (lf / 2) + (rf / 2); + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = SDL_SwapLE32((Uint32) (lf - ce)); + dst[3] = SDL_SwapLE32((Uint32) (rf - ce)); + dst[4] = SDL_SwapLE32((Uint32) ce); + dst[5] = SDL_SwapLE32((Uint32) ce); + } + } + } + break; + + case AUDIO_F32: + { + union { float f; Uint32 ui32; } f2i; /* !!! FIXME: lame. */ + float lf, rf, ce; + const Uint32 *src = (const Uint32 *) cvt->buf + cvt->len_cvt; + Uint32 *dst = (Uint32 *) cvt->buf + cvt->len_cvt * 3; + + if (SDL_AUDIO_ISBIGENDIAN(format)) { + for (i = cvt->len_cvt / 8; i; --i) { + dst -= 6; + src -= 2; + f2i.ui32 = SDL_SwapBE32(src[0]); + lf = f2i.f; + f2i.ui32 = SDL_SwapBE32(src[1]); + rf = f2i.f; + ce = (lf * 0.5f) + (rf * 0.5f); + dst[0] = src[0]; + dst[1] = src[1]; + f2i.f = (lf - ce); + dst[2] = SDL_SwapBE32(f2i.ui32); + f2i.f = (rf - ce); + dst[3] = SDL_SwapBE32(f2i.ui32); + f2i.f = ce; + f2i.ui32 = SDL_SwapBE32(f2i.ui32); + dst[4] = f2i.ui32; + dst[5] = f2i.ui32; + } + } else { + for (i = cvt->len_cvt / 8; i; --i) { + dst -= 6; + src -= 2; + f2i.ui32 = SDL_SwapLE32(src[0]); + lf = f2i.f; + f2i.ui32 = SDL_SwapLE32(src[1]); + rf = f2i.f; + ce = (lf * 0.5f) + (rf * 0.5f); + dst[0] = src[0]; + dst[1] = src[1]; + f2i.f = (lf - ce); + dst[2] = SDL_SwapLE32(f2i.ui32); + f2i.f = (rf - ce); + dst[3] = SDL_SwapLE32(f2i.ui32); + f2i.f = ce; + f2i.ui32 = SDL_SwapLE32(f2i.ui32); + dst[4] = f2i.ui32; + dst[5] = f2i.ui32; + } + } + } + break; + } cvt->len_cvt *= 3; if (cvt->filters[++cvt->filter_index]) { @@ -633,16 +631,16 @@ /* Duplicate a stereo channel to a pseudo-4.0 stream */ -void SDLCALL -SDL_ConvertSurround_4(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_ConvertSurround_4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; #ifdef DEBUG_CONVERT fprintf(stderr, "Converting stereo to quad\n"); #endif - switch (format & 0x8018) { + switch (format & (SDL_AUDIO_MASK_SIGNED|SDL_AUDIO_MASK_BITSIZE)) { case AUDIO_U8: { Uint8 *src, *dst, lf, rf, ce; @@ -691,7 +689,7 @@ src = cvt->buf + cvt->len_cvt; dst = cvt->buf + cvt->len_cvt * 2; - if ((format & 0x1000) == 0x1000) { + if (SDL_AUDIO_ISBIGENDIAN(format)) { for (i = cvt->len_cvt / 4; i; --i) { dst -= 8; src -= 4; @@ -741,7 +739,7 @@ src = cvt->buf + cvt->len_cvt; dst = cvt->buf + cvt->len_cvt * 2; - if ((format & 0x1000) == 0x1000) { + if (SDL_AUDIO_ISBIGENDIAN(format)) { for (i = cvt->len_cvt / 4; i; --i) { dst -= 8; src -= 4; @@ -782,218 +780,38 @@ } } break; - } - cvt->len_cvt *= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - -/* Convert 8-bit to 16-bit - LSB */ -void SDLCALL -SDL_Convert16LSB(SDL_AudioCVT * cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to 16-bit LSB\n"); -#endif - src = cvt->buf + cvt->len_cvt; - dst = cvt->buf + cvt->len_cvt * 2; - for (i = cvt->len_cvt; i; --i) { - src -= 1; - dst -= 2; - dst[1] = *src; - dst[0] = 0; - } - format = ((format & ~0x0008) | AUDIO_U16LSB); - cvt->len_cvt *= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - -/* Convert 8-bit to 16-bit - MSB */ -void SDLCALL -SDL_Convert16MSB(SDL_AudioCVT * cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to 16-bit MSB\n"); -#endif - src = cvt->buf + cvt->len_cvt; - dst = cvt->buf + cvt->len_cvt * 2; - for (i = cvt->len_cvt; i; --i) { - src -= 1; - dst -= 2; - dst[0] = *src; - dst[1] = 0; - } - format = ((format & ~0x0008) | AUDIO_U16MSB); - cvt->len_cvt *= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - -/* Convert 16-bit to 8-bit */ -void SDLCALL -SDL_Convert8(SDL_AudioCVT * cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting to 8-bit\n"); -#endif - src = cvt->buf; - dst = cvt->buf; - if ((format & 0x1000) != 0x1000) { /* Little endian */ - ++src; - } - for (i = cvt->len_cvt / 2; i; --i) { - *dst = *src; - src += 2; - dst += 1; - } - format = ((format & ~0x9010) | AUDIO_U8); - cvt->len_cvt /= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - -/* Toggle signed/unsigned */ -void SDLCALL -SDL_ConvertSign(SDL_AudioCVT * cvt, Uint16 format) -{ - int i; - Uint8 *data; + case AUDIO_S32: + { + const Uint32 *src = (const Uint32 *) (cvt->buf + cvt->len_cvt); + Uint32 *dst = (Uint32 *) (cvt->buf + cvt->len_cvt * 2); + Sint32 lf, rf, ce; -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio signedness\n"); -#endif - data = cvt->buf; - if ((format & 0xFF) == 16) { - if ((format & 0x1000) != 0x1000) { /* Little endian */ - ++data; - } - for (i = cvt->len_cvt / 2; i; --i) { - *data ^= 0x80; - data += 2; - } - } else { - for (i = cvt->len_cvt; i; --i) { - *data++ ^= 0x80; - } - } - format = (format ^ 0x8000); - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - -/* Toggle endianness */ -void SDLCALL -SDL_ConvertEndian(SDL_AudioCVT * cvt, Uint16 format) -{ - int i; - Uint8 *data, tmp; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio endianness\n"); -#endif - data = cvt->buf; - for (i = cvt->len_cvt / 2; i; --i) { - tmp = data[0]; - data[0] = data[1]; - data[1] = tmp; - data += 2; - } - format = (format ^ 0x1000); - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - -/* Convert rate up by multiple of 2 */ -void SDLCALL -SDL_RateMUL2(SDL_AudioCVT * cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); -#endif - src = cvt->buf + cvt->len_cvt; - dst = cvt->buf + cvt->len_cvt * 2; - switch (format & 0xFF) { - case 8: - for (i = cvt->len_cvt; i; --i) { - src -= 1; - dst -= 2; - dst[0] = src[0]; - dst[1] = src[0]; - } - break; - case 16: - for (i = cvt->len_cvt / 2; i; --i) { - src -= 2; - dst -= 4; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[0]; - dst[3] = src[1]; - } - break; - } - cvt->len_cvt *= 2; - if (cvt->filters[++cvt->filter_index]) { - cvt->filters[cvt->filter_index] (cvt, format); - } -} - - -/* Convert rate up by multiple of 2, for stereo */ -void SDLCALL -SDL_RateMUL2_c2(SDL_AudioCVT * cvt, Uint16 format) -{ - int i; - Uint8 *src, *dst; - -#ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); -#endif - src = cvt->buf + cvt->len_cvt; - dst = cvt->buf + cvt->len_cvt * 2; - switch (format & 0xFF) { - case 8: - for (i = cvt->len_cvt / 2; i; --i) { - src -= 2; - dst -= 4; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[0]; - dst[3] = src[1]; - } - break; - case 16: - for (i = cvt->len_cvt / 4; i; --i) { - src -= 4; - dst -= 8; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[0]; - dst[5] = src[1]; - dst[6] = src[2]; - dst[7] = src[3]; + if (SDL_AUDIO_ISBIGENDIAN(format)) { + for (i = cvt->len_cvt / 8; i; --i) { + dst -= 4; + src -= 2; + lf = (Sint32) SDL_SwapBE32(src[0]); + rf = (Sint32) SDL_SwapBE32(src[1]); + ce = (lf / 2) + (rf / 2); + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = SDL_SwapBE32((Uint32) (lf - ce)); + dst[3] = SDL_SwapBE32((Uint32) (rf - ce)); + } + } else { + for (i = cvt->len_cvt / 8; i; --i) { + dst -= 4; + src -= 2; + lf = (Sint32) SDL_SwapLE32(src[0]); + rf = (Sint32) SDL_SwapLE32(src[1]); + ce = (lf / 2) + (rf / 2); + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = SDL_SwapLE32((Uint32) (lf - ce)); + dst[3] = SDL_SwapLE32((Uint32) (rf - ce)); + } + } } break; } @@ -1003,56 +821,137 @@ } } -/* Convert rate up by multiple of 2, for quad */ -void SDLCALL -SDL_RateMUL2_c4(SDL_AudioCVT * cvt, Uint16 format) +/* Convert rate up by multiple of 2 */ +static void SDLCALL +SDL_RateMUL2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate * 2 (mono)\n"); +#endif + + #define mul2_mono(type) { \ + const type *src = (const type *) (cvt->buf + cvt->len_cvt); \ + type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \ + for (i = cvt->len_cvt / sizeof (type); i; --i) { \ + src--; \ + dst[-1] = dst[-2] = src[0]; \ + dst -= 2; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { + case 8: + mul2_mono(Uint8); + break; + case 16: + mul2_mono(Uint16); + break; + case 32: + mul2_mono(Uint32); + break; + } + + #undef mul2_mono + + cvt->len_cvt *= 2; + if (cvt->filters[++cvt->filter_index]) { + cvt->filters[cvt->filter_index] (cvt, format); + } +} + + +/* Convert rate up by multiple of 2, for stereo */ +static void SDLCALL +SDL_RateMUL2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format) +{ + int i; #ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); + fprintf(stderr, "Converting audio rate * 2 (stereo)\n"); #endif - src = cvt->buf + cvt->len_cvt; - dst = cvt->buf + cvt->len_cvt * 2; - switch (format & 0xFF) { + + #define mul2_stereo(type) { \ + const type *src = (const type *) (cvt->buf + cvt->len_cvt); \ + type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \ + for (i = cvt->len_cvt / (sizeof (type) * 2); i; --i) { \ + const type r = src[-1]; \ + const type l = src[-2]; \ + src -= 2; \ + dst[-1] = r; \ + dst[-2] = l; \ + dst[-3] = r; \ + dst[-4] = l; \ + dst -= 4; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { case 8: - for (i = cvt->len_cvt / 4; i; --i) { - src -= 4; - dst -= 8; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[0]; - dst[5] = src[1]; - dst[6] = src[2]; - dst[7] = src[3]; - } + mul2_stereo(Uint8); break; case 16: - for (i = cvt->len_cvt / 8; i; --i) { - src -= 8; - dst -= 16; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - dst[8] = src[0]; - dst[9] = src[1]; - dst[10] = src[2]; - dst[11] = src[3]; - dst[12] = src[4]; - dst[13] = src[5]; - dst[14] = src[6]; - dst[15] = src[7]; - } + mul2_stereo(Uint16); + break; + case 32: + mul2_stereo(Uint32); break; } + + #undef mul2_stereo + + cvt->len_cvt *= 2; + if (cvt->filters[++cvt->filter_index]) { + cvt->filters[cvt->filter_index] (cvt, format); + } +} + +/* Convert rate up by multiple of 2, for quad */ +static void SDLCALL +SDL_RateMUL2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format) +{ + int i; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate * 2 (quad)\n"); +#endif + + #define mul2_quad(type) { \ + const type *src = (const type *) (cvt->buf + cvt->len_cvt); \ + type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \ + for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \ + const type c1 = src[-1]; \ + const type c2 = src[-2]; \ + const type c3 = src[-3]; \ + const type c4 = src[-4]; \ + src -= 4; \ + dst[-1] = c1; \ + dst[-2] = c2; \ + dst[-3] = c3; \ + dst[-4] = c4; \ + dst[-5] = c1; \ + dst[-6] = c2; \ + dst[-7] = c3; \ + dst[-8] = c4; \ + dst -= 8; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { + case 8: + mul2_quad(Uint8); + break; + case 16: + mul2_quad(Uint16); + break; + case 32: + mul2_quad(Uint32); + break; + } + + #undef mul2_quad + cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -1061,67 +960,56 @@ /* Convert rate up by multiple of 2, for 5.1 */ -void SDLCALL -SDL_RateMUL2_c6(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_RateMUL2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Uint8 *src, *dst; #ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate * 2\n"); + fprintf(stderr, "Converting audio rate * 2 (six channels)\n"); #endif - src = cvt->buf + cvt->len_cvt; - dst = cvt->buf + cvt->len_cvt * 2; - switch (format & 0xFF) { + + #define mul2_chansix(type) { \ + const type *src = (const type *) (cvt->buf + cvt->len_cvt); \ + type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \ + for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \ + const type c1 = src[-1]; \ + const type c2 = src[-2]; \ + const type c3 = src[-3]; \ + const type c4 = src[-4]; \ + const type c5 = src[-5]; \ + const type c6 = src[-6]; \ + src -= 6; \ + dst[-1] = c1; \ + dst[-2] = c2; \ + dst[-3] = c3; \ + dst[-4] = c4; \ + dst[-5] = c5; \ + dst[-6] = c6; \ + dst[-7] = c1; \ + dst[-8] = c2; \ + dst[-9] = c3; \ + dst[-10] = c4; \ + dst[-11] = c5; \ + dst[-12] = c6; \ + dst -= 12; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { case 8: - for (i = cvt->len_cvt / 6; i; --i) { - src -= 6; - dst -= 12; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[0]; - dst[7] = src[1]; - dst[8] = src[2]; - dst[9] = src[3]; - dst[10] = src[4]; - dst[11] = src[5]; - } + mul2_chansix(Uint8); break; case 16: - for (i = cvt->len_cvt / 12; i; --i) { - src -= 12; - dst -= 24; - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - dst[8] = src[8]; - dst[9] = src[9]; - dst[10] = src[10]; - dst[11] = src[11]; - dst[12] = src[0]; - dst[13] = src[1]; - dst[14] = src[2]; - dst[15] = src[3]; - dst[16] = src[4]; - dst[17] = src[5]; - dst[18] = src[6]; - dst[19] = src[7]; - dst[20] = src[8]; - dst[21] = src[9]; - dst[22] = src[10]; - dst[23] = src[11]; - } + mul2_chansix(Uint16); + break; + case 32: + mul2_chansix(Uint32); break; } + + #undef mul2_chansix + cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -1129,34 +1017,39 @@ } /* Convert rate down by multiple of 2 */ -void SDLCALL -SDL_RateDIV2(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_RateDIV2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Uint8 *src, *dst; #ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); + fprintf(stderr, "Converting audio rate / 2 (mono)\n"); #endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { + + #define div2_mono(type) { \ + const type *src = (const type *) cvt->buf; \ + type *dst = (type *) cvt->buf; \ + for (i = cvt->len_cvt / (sizeof (type) * 2); i; --i) { \ + dst[0] = src[0]; \ + src += 2; \ + dst++; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { case 8: - for (i = cvt->len_cvt / 2; i; --i) { - dst[0] = src[0]; - src += 2; - dst += 1; - } + div2_mono(Uint8); break; case 16: - for (i = cvt->len_cvt / 4; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } + div2_mono(Uint16); + break; + case 32: + div2_mono(Uint32); break; } + + #undef div2_mono + cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -1165,37 +1058,40 @@ /* Convert rate down by multiple of 2, for stereo */ -void SDLCALL -SDL_RateDIV2_c2(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_RateDIV2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Uint8 *src, *dst; #ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); + fprintf(stderr, "Converting audio rate / 2 (stereo)\n"); #endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { + + #define div2_stereo(type) { \ + const type *src = (const type *) cvt->buf; \ + type *dst = (type *) cvt->buf; \ + for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \ + dst[0] = src[0]; \ + dst[1] = src[1]; \ + src += 4; \ + dst += 2; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { case 8: - for (i = cvt->len_cvt / 4; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - src += 4; - dst += 2; - } + div2_stereo(Uint8); break; case 16: - for (i = cvt->len_cvt / 8; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - src += 8; - dst += 4; - } + div2_stereo(Uint16); + break; + case 32: + div2_stereo(Uint32); break; } + + #undef div2_stereo + cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -1204,43 +1100,42 @@ /* Convert rate down by multiple of 2, for quad */ -void SDLCALL -SDL_RateDIV2_c4(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_RateDIV2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Uint8 *src, *dst; #ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); + fprintf(stderr, "Converting audio rate / 2 (quad)\n"); #endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { + + #define div2_quad(type) { \ + const type *src = (const type *) cvt->buf; \ + type *dst = (type *) cvt->buf; \ + for (i = cvt->len_cvt / (sizeof (type) * 8); i; --i) { \ + dst[0] = src[0]; \ + dst[1] = src[1]; \ + dst[2] = src[2]; \ + dst[3] = src[3]; \ + src += 8; \ + dst += 4; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { case 8: - for (i = cvt->len_cvt / 8; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - src += 8; - dst += 4; - } + div2_quad(Uint8); break; case 16: - for (i = cvt->len_cvt / 16; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - src += 16; - dst += 8; - } + div2_quad(Uint16); + break; + case 32: + div2_quad(Uint32); break; } + + #undef div2_quad + cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -1248,49 +1143,44 @@ } /* Convert rate down by multiple of 2, for 5.1 */ -void SDLCALL -SDL_RateDIV2_c6(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_RateDIV2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; - Uint8 *src, *dst; #ifdef DEBUG_CONVERT - fprintf(stderr, "Converting audio rate / 2\n"); + fprintf(stderr, "Converting audio rate / 2 (six channels)\n"); #endif - src = cvt->buf; - dst = cvt->buf; - switch (format & 0xFF) { + + #define div2_chansix(type) { \ + const type *src = (const type *) cvt->buf; \ + type *dst = (type *) cvt->buf; \ + for (i = cvt->len_cvt / (sizeof (type) * 12); i; --i) { \ + dst[0] = src[0]; \ + dst[1] = src[1]; \ + dst[2] = src[2]; \ + dst[3] = src[3]; \ + dst[4] = src[4]; \ + dst[5] = src[5]; \ + src += 12; \ + dst += 6; \ + } \ + } + + switch (SDL_AUDIO_BITSIZE(format)) { case 8: - for (i = cvt->len_cvt / 12; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - src += 12; - dst += 6; - } + div2_chansix(Uint8); break; case 16: - for (i = cvt->len_cvt / 24; i; --i) { - dst[0] = src[0]; - dst[1] = src[1]; - dst[2] = src[2]; - dst[3] = src[3]; - dst[4] = src[4]; - dst[5] = src[5]; - dst[6] = src[6]; - dst[7] = src[7]; - dst[8] = src[8]; - dst[9] = src[9]; - dst[10] = src[10]; - dst[11] = src[11]; - src += 24; - dst += 12; - } + div2_chansix(Uint16); + break; + case 32: + div2_chansix(Uint32); break; } + + #undef div_chansix + cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -1298,8 +1188,8 @@ } /* Very slow rate conversion routine */ -void SDLCALL -SDL_RateSLOW(SDL_AudioCVT * cvt, Uint16 format) +static void SDLCALL +SDL_RateSLOW(SDL_AudioCVT * cvt, SDL_AudioFormat format) { double ipos; int i, clen; @@ -1309,7 +1199,7 @@ #endif clen = (int) ((double) cvt->len_cvt / cvt->rate_incr); if (cvt->rate_incr > 1.0) { - switch (format & 0xFF) { + switch (SDL_AUDIO_BITSIZE(format)) { case 8: { Uint8 *output; @@ -1338,9 +1228,15 @@ } } break; + + case 32: + { + /* !!! FIXME: need 32-bit converter here! */ + fprintf(stderr, "FIXME: need 32-bit converter here!\n"); + } } } else { - switch (format & 0xFF) { + switch (SDL_AUDIO_BITSIZE(format)) { case 8: { Uint8 *output; @@ -1369,8 +1265,15 @@ } } break; + + case 32: + { + /* !!! FIXME: need 32-bit converter here! */ + fprintf(stderr, "FIXME: need 32-bit converter here!\n"); + } } } + cvt->len_cvt = clen; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); @@ -1397,57 +1300,106 @@ return (0); } -/* Creates a set of audio filters to convert from one format to another. - Returns -1 if the format conversion is not supported, or 1 if the - audio filter is set up. + +static SDL_AudioFilter +SDL_HandTunedTypeCVT(SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt) +{ + /* + * Fill in any future conversions that are specialized to a + * processor, platform, compiler, or library here. + */ + + return NULL; /* no specialized converter code available. */ +} + + +/* + * Find a converter between two data types. We try to select a hand-tuned + * asm/vectorized/optimized function first, and then fallback to an + * autogenerated function that is customized to convert between two + * specific data types. + */ +static int +SDL_BuildAudioTypeCVT(SDL_AudioCVT * cvt, + SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt) +{ + if (src_fmt != dst_fmt) { + const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt); + const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt); + SDL_AudioFilter filter = SDL_HandTunedTypeCVT(src_fmt, dst_fmt); + + /* No hand-tuned converter? Try the autogenerated ones. */ + if (filter == NULL) { + int i; + for (i = 0; sdl_audio_type_filters[i].filter != NULL; i++) { + const SDL_AudioTypeFilters *filt = &sdl_audio_type_filters[i]; + if ((filt->src_fmt == src_fmt) && (filt->dst_fmt == dst_fmt)) { + filter = filt->filter; + break; + } + } + + if (filter == NULL) { + return -1; /* Still no matching converter?! */ + } + } + + /* Update (cvt) with filter details... */ + cvt->filters[cvt->filter_index++] = filter; + if (src_bitsize < dst_bitsize) { + const int mult = (dst_bitsize / src_bitsize); + cvt->len_mult *= mult; + cvt->len_ratio *= mult; + } else if (src_bitsize > dst_bitsize) { + cvt->len_ratio /= (src_bitsize / dst_bitsize); + } + + return 1; /* added a converter. */ + } + + return 0; /* no conversion necessary. */ +} + + + +/* Creates a set of audio filters to convert from one format to another. + Returns -1 if the format conversion is not supported, 0 if there's + no conversion needed, or 1 if the audio filter is set up. */ int SDL_BuildAudioCVT(SDL_AudioCVT * cvt, - Uint16 src_format, Uint8 src_channels, int src_rate, - Uint16 dst_format, Uint8 dst_channels, int dst_rate) + SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate, + SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate) { -/*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n", - src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/ + /* there are no unsigned types over 16 bits, so catch this upfront. */ + if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) { + return -1; + } + if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) { + return -1; + } + + #ifdef DEBUG_CONVERT + printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n", + src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate); + #endif + /* Start off with no conversion necessary */ + + cvt->src_format = src_fmt; + cvt->dst_format = dst_fmt; cvt->needed = 0; cvt->filter_index = 0; cvt->filters[0] = NULL; cvt->len_mult = 1; cvt->len_ratio = 1.0; - /* First filter: Endian conversion from src to dst */ - if ((src_format & 0x1000) != (dst_format & 0x1000) - && ((src_format & 0xff) != 8)) { - cvt->filters[cvt->filter_index++] = SDL_ConvertEndian; - } - - /* Second filter: Sign conversion -- signed/unsigned */ - if ((src_format & 0x8000) != (dst_format & 0x8000)) { - cvt->filters[cvt->filter_index++] = SDL_ConvertSign; - } + /* Convert data types, if necessary. Updates (cvt). */ + if (SDL_BuildAudioTypeCVT(cvt, src_fmt, dst_fmt) == -1) + return -1; /* shouldn't happen, but just in case... */ - /* Next filter: Convert 16 bit <--> 8 bit PCM */ - if ((src_format & 0xFF) != (dst_format & 0xFF)) { - switch (dst_format & 0x10FF) { - case AUDIO_U8: - cvt->filters[cvt->filter_index++] = SDL_Convert8; - cvt->len_ratio /= 2; - break; - case AUDIO_U16LSB: - cvt->filters[cvt->filter_index++] = SDL_Convert16LSB; - cvt->len_mult *= 2; - cvt->len_ratio *= 2; - break; - case AUDIO_U16MSB: - cvt->filters[cvt->filter_index++] = SDL_Convert16MSB; - cvt->len_mult *= 2; - cvt->len_ratio *= 2; - break; - } - } - - /* Last filter: Mono/Stereo conversion */ + /* Channel conversion */ if (src_channels != dst_channels) { if ((src_channels == 1) && (dst_channels > 1)) { cvt->filters[cvt->filter_index++] = SDL_ConvertStereo; @@ -1504,7 +1456,7 @@ Uint32 hi_rate, lo_rate; int len_mult; double len_ratio; - void (SDLCALL * rate_cvt) (SDL_AudioCVT * cvt, Uint16 format); + SDL_AudioFilter rate_cvt = NULL; if (src_rate > dst_rate) { hi_rate = src_rate; @@ -1583,8 +1535,8 @@ /* Set up the filter information */ if (cvt->filter_index != 0) { cvt->needed = 1; - cvt->src_format = src_format; - cvt->dst_format = dst_format; + cvt->src_format = src_fmt; + cvt->dst_format = dst_fmt; cvt->len = 0; cvt->buf = NULL; cvt->filters[cvt->filter_index] = NULL;