sdl-ios-xcode: src/audio/SDL_audiocvt.c comparison

comparison 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

comparison

equal deleted inserted replaced

-:3f21778e7433
+:3b4ce57c6215
 #include "SDL_config.h"
 /* 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
+static void SDLCALL
-SDL_ConvertMono(SDL_AudioCVT * cvt, Uint16 format)
+SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
 {
 int i;
 Sint32 sample;
 #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;
 src = cvt->buf;
 {
 Uint8 *src, *dst;
 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]);
 if (sample > 65535) {
 dst[0] = 0xFF;
 {
 Uint8 *src, *dst;
 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]);
 if (sample > 32767) {
 dst[0] = 0x7F;
 dst += 2;
 }
 }
 }
 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
+static void SDLCALL
-SDL_ConvertStrip(SDL_AudioCVT * cvt, Uint16 format)
+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) {
+#define strip_chans_6_to_2(type) \
-case AUDIO_U8:
+{ \
-{
+const type *src = (const type *) cvt->buf; \
-Uint8 *src, *dst;
+type *dst = (type *) cvt->buf; \
+for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
-src = cvt->buf;
+dst[0] = src[0]; \
-dst = cvt->buf;
+dst[1] = src[1]; \
-for (i = cvt->len_cvt / 6; i; --i) {
+src += 6; \
-dst[0] = src[0];
+dst += 2; \
-dst[1] = src[1];
+} \
-src += 6;
+}
-dst += 2;
-}
+/* this function only cares about typesize, and data as a block of bits. */
-}
+switch (SDL_AUDIO_BITSIZE(format)) {
-break;
+case 8:
+strip_chans_6_to_2(Uint8);
-case AUDIO_S8:
+break;
-{
+case 16:
-Sint8 *src, *dst;
+strip_chans_6_to_2(Uint16);
+break;
-src = (Sint8 *) cvt->buf;
+case 32:
-dst = (Sint8 *) cvt->buf;
+strip_chans_6_to_2(Uint32);
-for (i = cvt->len_cvt / 6; i; --i) {
+break;
-dst[0] = src[0];
+}
-dst[1] = src[1];
-src += 6;
+#undef strip_chans_6_to_2
-dst += 2;
-}
-}
-break;
-case AUDIO_U16:
-{
-Uint8 *src, *dst;
-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;
-case AUDIO_S16:
-{
-Uint8 *src, *dst;
-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);
 }
 }
 /* Discard top 2 channels of 6 */
-void SDLCALL
+static void SDLCALL
-SDL_ConvertStrip_2(SDL_AudioCVT * cvt, Uint16 format)
+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) {
+#define strip_chans_6_to_4(type) \
-case AUDIO_U8:
+{ \
-{
+const type *src = (const type *) cvt->buf; \
-Uint8 *src, *dst;
+type *dst = (type *) cvt->buf; \
+for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
-src = cvt->buf;
+dst[0] = src[0]; \
-dst = cvt->buf;
+dst[1] = src[1]; \
-for (i = cvt->len_cvt / 4; i; --i) {
+dst[2] = src[2]; \
-dst[0] = src[0];
+dst[3] = src[3]; \
-dst[1] = src[1];
+src += 6; \
-src += 4;
+dst += 4; \
-dst += 2;
+} \
 }
-}
-break;
+/* this function only cares about typesize, and data as a block of bits. */
+switch (SDL_AUDIO_BITSIZE(format)) {
-case AUDIO_S8:
+case 8:
-{
+strip_chans_6_to_4(Uint8);
-Sint8 *src, *dst;
+break;
+case 16:
-src = (Sint8 *) cvt->buf;
+strip_chans_6_to_4(Uint16);
-dst = (Sint8 *) cvt->buf;
+break;
-for (i = cvt->len_cvt / 4; i; --i) {
+case 32:
-dst[0] = src[0];
+strip_chans_6_to_4(Uint32);
-dst[1] = src[1];
+break;
-src += 4;
+}
-dst += 2;
-}
+#undef strip_chans_6_to_4
-}
-break;
+cvt->len_cvt /= 6;
+cvt->len_cvt *= 4;
-case AUDIO_U16:
-{
-Uint8 *src, *dst;
-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;
-case AUDIO_S16:
-{
-Uint8 *src, *dst;
-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;
 if (cvt->filters[++cvt->filter_index]) {
 cvt->filters[cvt->filter_index] (cvt, format);
 }
 }
 /* Duplicate a mono channel to both stereo channels */
-void SDLCALL
+static void SDLCALL
-SDL_ConvertStereo(SDL_AudioCVT * cvt, Uint16 format)
+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) \
+{ \
-src = (Uint16 *) (cvt->buf + cvt->len_cvt);
+const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
-dst = (Uint16 *) (cvt->buf + cvt->len_cvt * 2);
+type *dst = (type *) (cvt->buf + cvt->len_cvt * 2); \
-for (i = cvt->len_cvt / 2; i; --i) {
+for (i = cvt->len_cvt / 2; i; --i, --src) { \
-dst -= 2;
+const type val = *src; \
-src -= 1;
+dst -= 2; \
-dst[0] = src[0];
+dst[0] = dst[1] = val; \
-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)) {
-src = cvt->buf + cvt->len_cvt;
+case 8:
-dst = cvt->buf + cvt->len_cvt * 2;
+dup_chans_1_to_2(Uint8);
-for (i = cvt->len_cvt; i; --i) {
+break;
-dst -= 2;
+case 16:
-src -= 1;
+dup_chans_1_to_2(Uint16);
-dst[0] = src[0];
+break;
-dst[1] = src[0];
+case 32:
-}
+dup_chans_1_to_2(Uint32);
-}
+break;
+}
+#undef dup_chans_1_to_2
 cvt->len_cvt *= 2;
 if (cvt->filters[++cvt->filter_index]) {
 cvt->filters[cvt->filter_index] (cvt, format);
 }
 }
 /* Duplicate a stereo channel to a pseudo-5.1 stream */
-void SDLCALL
+static void SDLCALL
-SDL_ConvertSurround(SDL_AudioCVT * cvt, Uint16 format)
+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;
 src = (Uint8 *) (cvt->buf + cvt->len_cvt);
 Uint16 lf, rf, ce, lr, rr;
 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;
 lf = (Uint16) ((src[0] << 8) | src[1]);
 rf = (Uint16) ((src[2] << 8) | src[3]);
 Sint16 lf, rf, ce, lr, rr;
 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;
 lf = (Sint16) ((src[0] << 8) | src[1]);
 rf = (Sint16) ((src[2] << 8) | src[3]);
 dst[3 + 8] = ((ce >> 8) & 0xFF);
 }
 }
 }
 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]) {
 cvt->filters[cvt->filter_index] (cvt, format);
 }
 }
 /* Duplicate a stereo channel to a pseudo-4.0 stream */
-void SDLCALL
+static void SDLCALL
-SDL_ConvertSurround_4(SDL_AudioCVT * cvt, Uint16 format)
+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;
 src = (Uint8 *) (cvt->buf + cvt->len_cvt);
 Uint16 lf, rf, ce, lr, rr;
 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;
 lf = (Uint16) ((src[0] << 8) | src[1]);
 rf = (Uint16) ((src[2] << 8) | src[3]);
 Sint16 lf, rf, ce, lr, rr;
 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;
 lf = (Sint16) ((src[0] << 8) | src[1]);
 rf = (Sint16) ((src[2] << 8) | src[3]);
 dst[3 + 4] = ((rr >> 8) & 0xFF);
 }
 }
 }
 break;
+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;
+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;
 }
 cvt->len_cvt *= 2;
 if (cvt->filters[++cvt->filter_index]) {
 cvt->filters[cvt->filter_index] (cvt, format);
 }
 }
+/* Convert rate up by multiple of 2 */
-/* Convert 8-bit to 16-bit - LSB */
+static void SDLCALL
-void SDLCALL
+SDL_RateMUL2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
-SDL_Convert16LSB(SDL_AudioCVT * cvt, Uint16 format)
 {
 int i;
-Uint8 *src, *dst;
 #ifdef DEBUG_CONVERT
-fprintf(stderr, "Converting to 16-bit LSB\n");
+fprintf(stderr, "Converting audio rate * 2 (mono)\n");
 #endif
-src = cvt->buf + cvt->len_cvt;
-dst = cvt->buf + cvt->len_cvt * 2;
+#define mul2_mono(type) { \
-for (i = cvt->len_cvt; i; --i) {
+const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
-src -= 1;
+type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
-dst -= 2;
+for (i = cvt->len_cvt / sizeof (type); i; --i) { \
-dst[1] = *src;
+src--; \
-dst[0] = 0;
+dst[-1] = dst[-2] = src[0]; \
-}
+dst -= 2; \
-format = ((format & ~0x0008) | AUDIO_U16LSB);
+} \
+}
+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 8-bit to 16-bit - MSB */
-void SDLCALL
+/* Convert rate up by multiple of 2, for stereo */
-SDL_Convert16MSB(SDL_AudioCVT * cvt, Uint16 format)
+static void SDLCALL
+SDL_RateMUL2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
 {
 int i;
-Uint8 *src, *dst;
 #ifdef DEBUG_CONVERT
-fprintf(stderr, "Converting to 16-bit MSB\n");
+fprintf(stderr, "Converting audio rate * 2 (stereo)\n");
 #endif
-src = cvt->buf + cvt->len_cvt;
-dst = cvt->buf + cvt->len_cvt * 2;
+#define mul2_stereo(type) { \
-for (i = cvt->len_cvt; i; --i) {
+const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
-src -= 1;
+type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
-dst -= 2;
+for (i = cvt->len_cvt / (sizeof (type) * 2); i; --i) { \
-dst[0] = *src;
+const type r = src[-1]; \
-dst[1] = 0;
+const type l = src[-2]; \
-}
+src -= 2; \
-format = ((format & ~0x0008) | AUDIO_U16MSB);
+dst[-1] = r; \
+dst[-2] = l; \
+dst[-3] = r; \
+dst[-4] = l; \
+dst -= 4; \
+} \
+}
+switch (SDL_AUDIO_BITSIZE(format)) {
+case 8:
+mul2_stereo(Uint8);
+break;
+case 16:
+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 16-bit to 8-bit */
+/* Convert rate up by multiple of 2, for quad */
-void SDLCALL
+static void SDLCALL
-SDL_Convert8(SDL_AudioCVT * cvt, Uint16 format)
+SDL_RateMUL2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
 {
 int i;
-Uint8 *src, *dst;
 #ifdef DEBUG_CONVERT
-fprintf(stderr, "Converting to 8-bit\n");
+fprintf(stderr, "Converting audio rate * 2 (quad)\n");
 #endif
-src = cvt->buf;
-dst = cvt->buf;
+#define mul2_quad(type) { \
-if ((format & 0x1000) != 0x1000) {  /* Little endian */
+const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
-++src;
+type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
-}
+for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \
-for (i = cvt->len_cvt / 2; i; --i) {
+const type c1 = src[-1]; \
-*dst = *src;
+const type c2 = src[-2]; \
-src += 2;
+const type c3 = src[-3]; \
-dst += 1;
+const type c4 = src[-4]; \
-}
+src -= 4; \
-format = ((format & ~0x9010) | AUDIO_U8);
+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);
+}
+}
+/* Convert rate up by multiple of 2, for 5.1 */
+static void SDLCALL
+SDL_RateMUL2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+int i;
+#ifdef DEBUG_CONVERT
+fprintf(stderr, "Converting audio rate * 2 (six channels)\n");
+#endif
+#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:
+mul2_chansix(Uint8);
+break;
+case 16:
+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);
+}
+}
+/* Convert rate down by multiple of 2 */
+static void SDLCALL
+SDL_RateDIV2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
+{
+int i;
+#ifdef DEBUG_CONVERT
+fprintf(stderr, "Converting audio rate / 2 (mono)\n");
+#endif
+#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:
+div2_mono(Uint8);
+break;
+case 16:
+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);
 }
 }
-/* Toggle signed/unsigned */
-void SDLCALL
+/* Convert rate down by multiple of 2, for stereo */
-SDL_ConvertSign(SDL_AudioCVT * cvt, Uint16 format)
+static void SDLCALL
+SDL_RateDIV2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
 {
 int i;
-Uint8 *data;
 #ifdef DEBUG_CONVERT
-fprintf(stderr, "Converting audio signedness\n");
+fprintf(stderr, "Converting audio rate / 2 (stereo)\n");
 #endif
-data = cvt->buf;
-if ((format & 0xFF) == 16) {
+#define div2_stereo(type) { \
-if ((format & 0x1000) != 0x1000) {      /* Little endian */
+const type *src = (const type *) cvt->buf; \
-++data;
+type *dst = (type *) cvt->buf; \
-}
+for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \
-for (i = cvt->len_cvt / 2; i; --i) {
+dst[0] = src[0]; \
-*data ^= 0x80;
+dst[1] = src[1]; \
-data += 2;
+src += 4; \
-}
+dst += 2; \
-} else {
+} \
-for (i = cvt->len_cvt; i; --i) {
+}
-*data++ ^= 0x80;
-}
+switch (SDL_AUDIO_BITSIZE(format)) {
-}
-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) {
+div2_stereo(Uint8);
-src -= 1;
-dst -= 2;
-dst[0] = src[0];
-dst[1] = src[0];
-}
 break;
 case 16:
-for (i = cvt->len_cvt / 2; i; --i) {
+div2_stereo(Uint16);
-src -= 2;
+break;
-dst -= 4;
+case 32:
-dst[0] = src[0];
+div2_stereo(Uint32);
-dst[1] = src[1];
+break;
-dst[2] = src[0];
+}
-dst[3] = src[1];
-}
+#undef div2_stereo
-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];
-}
-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 quad */
-void SDLCALL
-SDL_RateMUL2_c4(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 / 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];
-}
-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];
-}
-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 5.1 */
-void SDLCALL
-SDL_RateMUL2_c6(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 / 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];
-}
-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];
-}
-break;
-}
-cvt->len_cvt *= 2;
-if (cvt->filters[++cvt->filter_index]) {
-cvt->filters[cvt->filter_index] (cvt, format);
-}
-}
-/* Convert rate down by multiple of 2 */
-void SDLCALL
-SDL_RateDIV2(SDL_AudioCVT * cvt, Uint16 format)
-{
-int i;
-Uint8 *src, *dst;
-#ifdef DEBUG_CONVERT
-fprintf(stderr, "Converting audio rate / 2\n");
-#endif
-src = cvt->buf;
-dst = cvt->buf;
-switch (format & 0xFF) {
-case 8:
-for (i = cvt->len_cvt / 2; i; --i) {
-dst[0] = src[0];
-src += 2;
-dst += 1;
-}
-break;
-case 16:
-for (i = cvt->len_cvt / 4; i; --i) {
-dst[0] = src[0];
-dst[1] = src[1];
-src += 4;
-dst += 2;
-}
-break;
-}
 cvt->len_cvt /= 2;
 if (cvt->filters[++cvt->filter_index]) {
 cvt->filters[cvt->filter_index] (cvt, format);
 }
 }
-/* Convert rate down by multiple of 2, for stereo */
+/* Convert rate down by multiple of 2, for quad */
-void SDLCALL
+static void SDLCALL
-SDL_RateDIV2_c2(SDL_AudioCVT * cvt, Uint16 format)
+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;
+#define div2_quad(type) { \
-switch (format & 0xFF) {
+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 / 4; i; --i) {
+div2_quad(Uint8);
-dst[0] = src[0];
-dst[1] = src[1];
-src += 4;
-dst += 2;
-}
 break;
 case 16:
-for (i = cvt->len_cvt / 8; i; --i) {
+div2_quad(Uint16);
-dst[0] = src[0];
+break;
-dst[1] = src[1];
+case 32:
-dst[2] = src[2];
+div2_quad(Uint32);
-dst[3] = src[3];
+break;
-src += 8;
+}
-dst += 4;
-}
+#undef div2_quad
-break;
-}
 cvt->len_cvt /= 2;
 if (cvt->filters[++cvt->filter_index]) {
 cvt->filters[cvt->filter_index] (cvt, format);
 }
 }
+/* Convert rate down by multiple of 2, for 5.1 */
-/* Convert rate down by multiple of 2, for quad */
+static void SDLCALL
-void SDLCALL
+SDL_RateDIV2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format)
-SDL_RateDIV2_c4(SDL_AudioCVT * cvt, Uint16 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;
+#define div2_chansix(type) { \
-switch (format & 0xFF) {
+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 / 8; i; --i) {
+div2_chansix(Uint8);
-dst[0] = src[0];
-dst[1] = src[1];
-dst[2] = src[2];
-dst[3] = src[3];
-src += 8;
-dst += 4;
-}
 break;
 case 16:
-for (i = cvt->len_cvt / 16; i; --i) {
+div2_chansix(Uint16);
-dst[0] = src[0];
+break;
-dst[1] = src[1];
+case 32:
-dst[2] = src[2];
+div2_chansix(Uint32);
-dst[3] = src[3];
+break;
-dst[4] = src[4];
+}
-dst[5] = src[5];
-dst[6] = src[6];
+#undef div_chansix
-dst[7] = src[7];
-src += 16;
-dst += 8;
-}
-break;
-}
 cvt->len_cvt /= 2;
 if (cvt->filters[++cvt->filter_index]) {
 cvt->filters[cvt->filter_index] (cvt, format);
 }
 }
-/* Convert rate down by multiple of 2, for 5.1 */
-void SDLCALL
-SDL_RateDIV2_c6(SDL_AudioCVT * cvt, Uint16 format)
-{
-int i;
-Uint8 *src, *dst;
-#ifdef DEBUG_CONVERT
-fprintf(stderr, "Converting audio rate / 2\n");
-#endif
-src = cvt->buf;
-dst = cvt->buf;
-switch (format & 0xFF) {
-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;
-}
-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;
-}
-break;
-}
-cvt->len_cvt /= 2;
-if (cvt->filters[++cvt->filter_index]) {
-cvt->filters[cvt->filter_index] (cvt, format);
-}
-}
 /* Very slow rate conversion routine */
-void SDLCALL
+static void SDLCALL
-SDL_RateSLOW(SDL_AudioCVT * cvt, Uint16 format)
+SDL_RateSLOW(SDL_AudioCVT * cvt, SDL_AudioFormat format)
 {
 double ipos;
 int i, clen;
 #ifdef DEBUG_CONVERT
 fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0 / cvt->rate_incr);
 #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;
 output = cvt->buf;
 ipos += cvt->rate_incr;
 output += 1;
 }
 }
 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;
 output = cvt->buf + clen;
 output -= 1;
 *output = ((Uint16 *) cvt->buf)[(int) ipos];
 }
 }
 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);
 }
 }
 cvt->filter_index = 0;
 cvt->filters[0] (cvt, cvt->src_format);
 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
+static SDL_AudioFilter
-audio filter is set up.
+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,
+SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
-Uint16 dst_format, Uint8 dst_channels, int dst_rate)
+SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
 {
-/*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
+/* there are no unsigned types over 16 bits, so catch this upfront. */
-		src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/
+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 */
+/* Convert data types, if necessary. Updates (cvt). */
-if ((src_format & 0x1000) != (dst_format & 0x1000)
+if (SDL_BuildAudioTypeCVT(cvt, src_fmt, dst_fmt) == -1)
-&& ((src_format & 0xff) != 8)) {
+return -1;  /* shouldn't happen, but just in case... */
-cvt->filters[cvt->filter_index++] = SDL_ConvertEndian;
-}
+/* Channel conversion */
-/* Second filter: Sign conversion -- signed/unsigned */
-if ((src_format & 0x8000) != (dst_format & 0x8000)) {
-cvt->filters[cvt->filter_index++] = SDL_ConvertSign;
-}
-/* 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 */
 if (src_channels != dst_channels) {
 if ((src_channels == 1) && (dst_channels > 1)) {
 cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
 cvt->len_mult *= 2;
 src_channels = 2;
 cvt->rate_incr = 0.0;
 if ((src_rate / 100) != (dst_rate / 100)) {
 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;
 lo_rate = dst_rate;
 switch (src_channels) {
 }
 /* Set up the filter information */
 if (cvt->filter_index != 0) {
 cvt->needed = 1;
-cvt->src_format = src_format;
+cvt->src_format = src_fmt;
-cvt->dst_format = dst_format;
+cvt->dst_format = dst_fmt;
 cvt->len = 0;
 cvt->buf = NULL;
 cvt->filters[cvt->filter_index] = NULL;
 }
 return (cvt->needed);

Mercurial > sdl-ios-xcode

comparison src/audio/SDL_audiocvt.c @ 1982:3b4ce57c6215