frameworks/SDL2.git - OutOfOrder git server

			@@ -1,6 +1,6 @@
			/*
			Simple DirectMedia Layer
			Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
			Copyright (C) 1997-2020 Sam Lantinga <slouken@libsdl.org>

			This software is provided 'as-is', without any express or implied
			warranty. In no event will the authors be held liable for any damages
			@@ -20,248 +20,856 @@
			*/
			#include "../SDL_internal.h"

			#ifdef HAVE_LIMITS_H
			#include <limits.h>
			#else
			#ifndef SIZE_MAX
			#define SIZE_MAX ((size_t)-1)
			#endif
			#ifndef INT_MAX
			/* Make a lucky guess. */
			#define INT_MAX SDL_MAX_SINT32
			#endif
			#endif

			/* Microsoft WAVE file loading routines */

			#include "SDL_log.h"
			#include "SDL_hints.h"
			#include "SDL_audio.h"
			#include "SDL_wave.h"
			#include "SDL_audio_c.h"


			static int ReadChunk(SDL_RWops * src, Chunk * chunk);

			struct MS_ADPCM_decodestate
			/* Reads the value stored at the location of the f1 pointer, multiplies it
			* with the second argument and then stores the result to f1.
			* Returns 0 on success, or -1 if the multiplication overflows, in which case f1
			* does not get modified.
			*/
			static int
			SafeMult(size_t *f1, size_t f2)
			{
			Uint8 hPredictor;
			Uint16 iDelta;
			Sint16 iSamp1;
			Sint16 iSamp2;
			};
			static struct MS_ADPCM_decoder
			if (f1 > 0 && SIZE_MAX / f1 <= f2) {
			return -1;
			}
			f1 = f2;
			return 0;
			}

			typedef struct ADPCM_DecoderState
			{
			WaveFMT wavefmt;
			Uint16 wSamplesPerBlock;
			Uint16 wNumCoef;
			Sint16 aCoeff[7][2];
			/* * * */
			struct MS_ADPCM_decodestate state[2];
			} MS_ADPCM_state;
			Uint32 channels; /* Number of channels. */
			size_t blocksize; /* Size of an ADPCM block in bytes. */
			size_t blockheadersize; /* Size of an ADPCM block header in bytes. */
			size_t samplesperblock; /* Number of samples per channel in an ADPCM block. */
			size_t framesize; /* Size of a sample frame (16-bit PCM) in bytes. */
			Sint64 framestotal; /* Total number of sample frames. */
			Sint64 framesleft; /* Number of sample frames still to be decoded. */
			void ddata; / Decoder data from initialization. */
			void cstate; / Decoding state for each channel. */

			/* ADPCM data. */
			struct {
			Uint8 *data;
			size_t size;
			size_t pos;
			} input;

			/* Current ADPCM block in the ADPCM data above. */
			struct {
			Uint8 *data;
			size_t size;
			size_t pos;
			} block;

			/* Decoded 16-bit PCM data. */
			struct {
			Sint16 *data;
			size_t size;
			size_t pos;
			} output;
			} ADPCM_DecoderState;

			typedef struct MS_ADPCM_CoeffData
			{
			Uint16 coeffcount;
			Sint16 *coeff;
			Sint16 aligndummy; /* Has to be last member. */
			} MS_ADPCM_CoeffData;

			typedef struct MS_ADPCM_ChannelState
			{
			Uint16 delta;
			Sint16 coeff1;
			Sint16 coeff2;
			} MS_ADPCM_ChannelState;

			#ifdef SDL_WAVE_DEBUG_LOG_FORMAT
			static void
			WaveDebugLogFormat(WaveFile *file)
			{
			WaveFormat *format = &file->format;
			const char *fmtstr = "WAVE file: %s, %u Hz, %s, %u bits, %u %s/s";
			const char waveformat, wavechannel, *wavebpsunit = "B";
			Uint32 wavebps = format->byterate;
			char channelstr[64];

			SDL_zeroa(channelstr);

			switch (format->encoding) {
			case PCM_CODE:
			waveformat = "PCM";
			break;
			case IEEE_FLOAT_CODE:
			waveformat = "IEEE Float";
			break;
			case ALAW_CODE:
			waveformat = "A-law";
			break;
			case MULAW_CODE:
			waveformat = "\xc2\xb5-law";
			break;
			case MS_ADPCM_CODE:
			waveformat = "MS ADPCM";
			break;
			case IMA_ADPCM_CODE:
			waveformat = "IMA ADPCM";
			break;
			default:
			waveformat = "Unknown";
			break;
			}

			#define SDL_WAVE_DEBUG_CHANNELCFG(STR, CODE) case CODE: wavechannel = STR; break;
			#define SDL_WAVE_DEBUG_CHANNELSTR(STR, CODE) if (format->channelmask & CODE) { \
			SDL_strlcat(channelstr, channelstr[0] ? "-" STR : STR, sizeof(channelstr));}

			if (format->formattag == EXTENSIBLE_CODE && format->channelmask > 0) {
			switch (format->channelmask) {
			SDL_WAVE_DEBUG_CHANNELCFG("1.0 Mono", 0x4)
			SDL_WAVE_DEBUG_CHANNELCFG("1.1 Mono", 0xc)
			SDL_WAVE_DEBUG_CHANNELCFG("2.0 Stereo", 0x3)
			SDL_WAVE_DEBUG_CHANNELCFG("2.1 Stereo", 0xb)
			SDL_WAVE_DEBUG_CHANNELCFG("3.0 Stereo", 0x7)
			SDL_WAVE_DEBUG_CHANNELCFG("3.1 Stereo", 0xf)
			SDL_WAVE_DEBUG_CHANNELCFG("3.0 Surround", 0x103)
			SDL_WAVE_DEBUG_CHANNELCFG("3.1 Surround", 0x10b)
			SDL_WAVE_DEBUG_CHANNELCFG("4.0 Quad", 0x33)
			SDL_WAVE_DEBUG_CHANNELCFG("4.1 Quad", 0x3b)
			SDL_WAVE_DEBUG_CHANNELCFG("4.0 Surround", 0x107)
			SDL_WAVE_DEBUG_CHANNELCFG("4.1 Surround", 0x10f)
			SDL_WAVE_DEBUG_CHANNELCFG("5.0", 0x37)
			SDL_WAVE_DEBUG_CHANNELCFG("5.1", 0x3f)
			SDL_WAVE_DEBUG_CHANNELCFG("5.0 Side", 0x607)
			SDL_WAVE_DEBUG_CHANNELCFG("5.1 Side", 0x60f)
			SDL_WAVE_DEBUG_CHANNELCFG("6.0", 0x137)
			SDL_WAVE_DEBUG_CHANNELCFG("6.1", 0x13f)
			SDL_WAVE_DEBUG_CHANNELCFG("6.0 Side", 0x707)
			SDL_WAVE_DEBUG_CHANNELCFG("6.1 Side", 0x70f)
			SDL_WAVE_DEBUG_CHANNELCFG("7.0", 0xf7)
			SDL_WAVE_DEBUG_CHANNELCFG("7.1", 0xff)
			SDL_WAVE_DEBUG_CHANNELCFG("7.0 Side", 0x6c7)
			SDL_WAVE_DEBUG_CHANNELCFG("7.1 Side", 0x6cf)
			SDL_WAVE_DEBUG_CHANNELCFG("7.0 Surround", 0x637)
			SDL_WAVE_DEBUG_CHANNELCFG("7.1 Surround", 0x63f)
			SDL_WAVE_DEBUG_CHANNELCFG("9.0 Surround", 0x5637)
			SDL_WAVE_DEBUG_CHANNELCFG("9.1 Surround", 0x563f)
			SDL_WAVE_DEBUG_CHANNELCFG("11.0 Surround", 0x56f7)
			SDL_WAVE_DEBUG_CHANNELCFG("11.1 Surround", 0x56ff)
			default:
			SDL_WAVE_DEBUG_CHANNELSTR("FL", 0x1)
			SDL_WAVE_DEBUG_CHANNELSTR("FR", 0x2)
			SDL_WAVE_DEBUG_CHANNELSTR("FC", 0x4)
			SDL_WAVE_DEBUG_CHANNELSTR("LF", 0x8)
			SDL_WAVE_DEBUG_CHANNELSTR("BL", 0x10)
			SDL_WAVE_DEBUG_CHANNELSTR("BR", 0x20)
			SDL_WAVE_DEBUG_CHANNELSTR("FLC", 0x40)
			SDL_WAVE_DEBUG_CHANNELSTR("FRC", 0x80)
			SDL_WAVE_DEBUG_CHANNELSTR("BC", 0x100)
			SDL_WAVE_DEBUG_CHANNELSTR("SL", 0x200)
			SDL_WAVE_DEBUG_CHANNELSTR("SR", 0x400)
			SDL_WAVE_DEBUG_CHANNELSTR("TC", 0x800)
			SDL_WAVE_DEBUG_CHANNELSTR("TFL", 0x1000)
			SDL_WAVE_DEBUG_CHANNELSTR("TFC", 0x2000)
			SDL_WAVE_DEBUG_CHANNELSTR("TFR", 0x4000)
			SDL_WAVE_DEBUG_CHANNELSTR("TBL", 0x8000)
			SDL_WAVE_DEBUG_CHANNELSTR("TBC", 0x10000)
			SDL_WAVE_DEBUG_CHANNELSTR("TBR", 0x20000)
			break;
			}
			} else {
			switch (format->channels) {
			default:
			if (SDL_snprintf(channelstr, sizeof(channelstr), "%u channels", format->channels) >= 0) {
			wavechannel = channelstr;
			break;
			}
			case 0:
			wavechannel = "Unknown";
			break;
			case 1:
			wavechannel = "Mono";
			break;
			case 2:
			wavechannel = "Setero";
			break;
			}
			}

			#undef SDL_WAVE_DEBUG_CHANNELCFG
			#undef SDL_WAVE_DEBUG_CHANNELSTR

			if (wavebps >= 1024) {
			wavebpsunit = "KiB";
			wavebps = wavebps / 1024 + (wavebps & 0x3ff ? 1 : 0);
			}

			SDL_LogDebug(SDL_LOG_CATEGORY_AUDIO, fmtstr, waveformat, format->frequency, wavechannel, format->bitspersample, wavebps, wavebpsunit);
			}
			#endif

			#ifdef SDL_WAVE_DEBUG_DUMP_FORMAT
			static void
			WaveDebugDumpFormat(WaveFile *file, Uint32 rifflen, Uint32 fmtlen, Uint32 datalen)
			{
			WaveFormat *format = &file->format;
			const char *fmtstr1 = "WAVE chunk dump:\n"
			"-------------------------------------------\n"
			"RIFF %11u\n"
			"-------------------------------------------\n"
			" fmt %11u\n"
			" wFormatTag 0x%04x\n"
			" nChannels %11u\n"
			" nSamplesPerSec %11u\n"
			" nAvgBytesPerSec %11u\n"
			" nBlockAlign %11u\n";
			const char *fmtstr2 = " wBitsPerSample %11u\n";
			const char *fmtstr3 = " cbSize %11u\n";
			const char *fmtstr4a = " wValidBitsPerSample %11u\n";
			const char *fmtstr4b = " wSamplesPerBlock %11u\n";
			const char *fmtstr5 = " dwChannelMask 0x%08x\n"
			" SubFormat\n"
			" %08x-%04x-%04x-%02x%02x%02x%02x%02x%02x%02x%02x\n";
			const char *fmtstr6 = "-------------------------------------------\n"
			" fact\n"
			" dwSampleLength %11u\n";
			const char *fmtstr7 = "-------------------------------------------\n"
			" data %11u\n"
			"-------------------------------------------\n";
			char *dumpstr;
			size_t dumppos = 0;
			const size_t bufsize = 1024;
			int res;

			dumpstr = SDL_malloc(bufsize);
			if (dumpstr == NULL) {
			return;
			}
			dumpstr[0] = 0;

			res = SDL_snprintf(dumpstr, bufsize, fmtstr1, rifflen, fmtlen, format->formattag, format->channels, format->frequency, format->byterate, format->blockalign);
			dumppos += res > 0 ? res : 0;
			if (fmtlen >= 16) {
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr2, format->bitspersample);
			dumppos += res > 0 ? res : 0;
			}
			if (fmtlen >= 18) {
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr3, format->extsize);
			dumppos += res > 0 ? res : 0;
			}
			if (format->formattag == EXTENSIBLE_CODE && fmtlen >= 40 && format->extsize >= 22) {
			const Uint8 *g = format->subformat;
			const Uint32 g1 = g[0] \| ((Uint32)g[1] << 8) \| ((Uint32)g[2] << 16) \| ((Uint32)g[3] << 24);
			const Uint32 g2 = g[4] \| ((Uint32)g[5] << 8);
			const Uint32 g3 = g[6] \| ((Uint32)g[7] << 8);

			switch (format->encoding) {
			default:
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr4a, format->validsamplebits);
			dumppos += res > 0 ? res : 0;
			break;
			case MS_ADPCM_CODE:
			case IMA_ADPCM_CODE:
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr4b, format->samplesperblock);
			dumppos += res > 0 ? res : 0;
			break;
			}
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr5, format->channelmask, g1, g2, g3, g[8], g[9], g[10], g[11], g[12], g[13], g[14], g[15]);
			dumppos += res > 0 ? res : 0;
			} else {
			switch (format->encoding) {
			case MS_ADPCM_CODE:
			case IMA_ADPCM_CODE:
			if (fmtlen >= 20 && format->extsize >= 2) {
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr4b, format->samplesperblock);
			dumppos += res > 0 ? res : 0;
			}
			break;
			}
			}
			if (file->fact.status >= 1) {
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr6, file->fact.samplelength);
			dumppos += res > 0 ? res : 0;
			}
			res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr7, datalen);
			dumppos += res > 0 ? res : 0;

			SDL_LogDebug(SDL_LOG_CATEGORY_AUDIO, "%s", dumpstr);

			free(dumpstr);
			}
			#endif

			static Sint64
			WaveAdjustToFactValue(WaveFile *file, Sint64 sampleframes)
			{
			if (file->fact.status == 2) {
			if (file->facthint == FactStrict && sampleframes < file->fact.samplelength) {
			return SDL_SetError("Invalid number of sample frames in WAVE fact chunk (too many)");
			} else if (sampleframes > file->fact.samplelength) {
			return file->fact.samplelength;
			}
			}

			return sampleframes;
			}

			static int
			InitMS_ADPCM(WaveFMT * format)
			MS_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
			{
			Uint8 *rogue_feel;
			int i;
			WaveFormat *format = &file->format;
			const size_t blockheadersize = (size_t)file->format.channels * 7;
			const size_t availableblocks = datalength / file->format.blockalign;
			const size_t blockframebitsize = (size_t)file->format.bitspersample * file->format.channels;
			const size_t trailingdata = datalength % file->format.blockalign;

			/* Set the rogue pointer to the MS_ADPCM specific data */
			MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
			MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
			MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
			MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
			MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
			MS_ADPCM_state.wavefmt.bitspersample =
			SDL_SwapLE16(format->bitspersample);
			rogue_feel = (Uint8 ) format + sizeof(format);
			if (sizeof(*format) == 16) {
			/* const Uint16 extra_info = ((rogue_feel[1] << 8) \| rogue_feel[0]); */
			rogue_feel += sizeof(Uint16);
			if (file->trunchint == TruncVeryStrict \|\| file->trunchint == TruncStrict) {
			/* The size of the data chunk must be a multiple of the block size. */
			if (datalength < blockheadersize \|\| trailingdata > 0) {
			return SDL_SetError("Truncated MS ADPCM block");
			}
			MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1] << 8) \| rogue_feel[0]);
			rogue_feel += sizeof(Uint16);
			MS_ADPCM_state.wNumCoef = ((rogue_feel[1] << 8) \| rogue_feel[0]);
			rogue_feel += sizeof(Uint16);
			if (MS_ADPCM_state.wNumCoef != 7) {
			SDL_SetError("Unknown set of MS_ADPCM coefficients");
			return (-1);
			}
			for (i = 0; i < MS_ADPCM_state.wNumCoef; ++i) {
			MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1] << 8) \| rogue_feel[0]);
			rogue_feel += sizeof(Uint16);
			MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1] << 8) \| rogue_feel[0]);
			rogue_feel += sizeof(Uint16);
			}
			return (0);
			}

			static Sint32
			MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,
			Uint8 nybble, Sint16 * coeff)
			/* Calculate number of sample frames that will be decoded. */
			file->sampleframes = (Sint64)availableblocks * format->samplesperblock;
			if (trailingdata > 0) {
			/* The last block is truncated. Check if we can get any samples out of it. */
			if (file->trunchint == TruncDropFrame) {
			/* Drop incomplete sample frame. */
			if (trailingdata >= blockheadersize) {
			size_t trailingsamples = 2 + (trailingdata - blockheadersize) * 8 / blockframebitsize;
			if (trailingsamples > format->samplesperblock) {
			trailingsamples = format->samplesperblock;
			}
			file->sampleframes += trailingsamples;
			}
			}
			}

			file->sampleframes = WaveAdjustToFactValue(file, file->sampleframes);
			if (file->sampleframes < 0) {
			return -1;
			}

			return 0;
			}

			static int
			MS_ADPCM_Init(WaveFile *file, size_t datalength)
			{
			const Sint32 max_audioval = ((1 << (16 - 1)) - 1);
			const Sint32 min_audioval = -(1 << (16 - 1));
			const Sint32 adaptive[] = {
			WaveFormat *format = &file->format;
			WaveChunk *chunk = &file->chunk;
			const size_t blockheadersize = (size_t)format->channels * 7;
			const size_t blockdatasize = (size_t)format->blockalign - blockheadersize;
			const size_t blockframebitsize = (size_t)format->bitspersample * format->channels;
			const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize;
			const Sint16 presetcoeffs[14] = {256, 0, 512, -256, 0, 0, 192, 64, 240, 0, 460, -208, 392, -232};
			size_t i, coeffcount;
			MS_ADPCM_CoeffData *coeffdata;

			/* Sanity checks. */

			/* While it's clear how IMA ADPCM handles more than two channels, the nibble
			* order of MS ADPCM makes it awkward. The Standards Update does not talk
			* about supporting more than stereo anyway.
			*/
			if (format->channels > 2) {
			return SDL_SetError("Invalid number of channels");
			}

			if (format->bitspersample != 4) {
			return SDL_SetError("Invalid MS ADPCM bits per sample of %u", (unsigned int)format->bitspersample);
			}

			/* The block size must be big enough to contain the block header. */
			if (format->blockalign < blockheadersize) {
			return SDL_SetError("Invalid MS ADPCM block size (nBlockAlign)");
			}

			if (format->formattag == EXTENSIBLE_CODE) {
			/* Does have a GUID (like all format tags), but there's no specification
			* for how the data is packed into the extensible header. Making
			* assumptions here could lead to new formats nobody wants to support.
			*/
			return SDL_SetError("MS ADPCM with the extensible header is not supported");
			}

			/* There are wSamplesPerBlock, wNumCoef, and at least 7 coefficient pairs in
			* the extended part of the header.
			*/
			if (chunk->size < 22) {
			return SDL_SetError("Could not read MS ADPCM format header");
			}

			format->samplesperblock = chunk->data[18] \| ((Uint16)chunk->data[19] << 8);
			/* Number of coefficient pairs. A pair has two 16-bit integers. */
			coeffcount = chunk->data[20] \| ((size_t)chunk->data[21] << 8);
			/* bPredictor, the integer offset into the coefficients array, is only
			* 8 bits. It can only address the first 256 coefficients. Let's limit
			* the count number here.
			*/
			if (coeffcount > 256) {
			coeffcount = 256;
			}

			if (chunk->size < 22 + coeffcount * 4) {
			return SDL_SetError("Could not read custom coefficients in MS ADPCM format header");
			} else if (format->extsize < 4 + coeffcount * 4) {
			return SDL_SetError("Invalid MS ADPCM format header (too small)");
			} else if (coeffcount < 7) {
			return SDL_SetError("Missing required coefficients in MS ADPCM format header");
			}

			coeffdata = (MS_ADPCM_CoeffData )SDL_malloc(sizeof(MS_ADPCM_CoeffData) + coeffcount 4);
			file->decoderdata = coeffdata; /* Freed in cleanup. */
			if (coeffdata == NULL) {
			return SDL_OutOfMemory();
			}
			coeffdata->coeff = &coeffdata->aligndummy;
			coeffdata->coeffcount = (Uint16)coeffcount;

			/* Copy the 16-bit pairs. */
			for (i = 0; i < coeffcount * 2; i++) {
			Sint32 c = chunk->data[22 + i * 2] \| ((Sint32)chunk->data[23 + i * 2] << 8);
			if (c >= 0x8000) {
			c -= 0x10000;
			}
			if (i < 14 && c != presetcoeffs[i]) {
			return SDL_SetError("Wrong preset coefficients in MS ADPCM format header");
			}
			coeffdata->coeff[i] = (Sint16)c;
			}

			/* Technically, wSamplesPerBlock is required, but we have all the
			* information in the other fields to calculate it, if it's zero.
			*/
			if (format->samplesperblock == 0) {
			/* Let's be nice to the encoders that didn't know how to fill this.
			* The Standards Update calculates it this way:
			*
			* x = Block size (in bits) minus header size (in bits)
			* y = Bit depth multiplied by channel count
			* z = Number of samples per channel in block header
			* wSamplesPerBlock = x / y + z
			*/
			format->samplesperblock = (Uint32)blockdatasamples + 2;
			}

			/* nBlockAlign can be in conflict with wSamplesPerBlock. For example, if
			* the number of samples doesn't fit into the block. The Standards Update
			* also describes wSamplesPerBlock with a formula that makes it necessary to
			* always fill the block with the maximum amount of samples, but this is not
			* enforced here as there are no compatibility issues.
			* A truncated block header with just one sample is not supported.
			*/
			if (format->samplesperblock == 1 \|\| blockdatasamples < format->samplesperblock - 2) {
			return SDL_SetError("Invalid number of samples per MS ADPCM block (wSamplesPerBlock)");
			}

			if (MS_ADPCM_CalculateSampleFrames(file, datalength) < 0) {
			return -1;
			}

			return 0;
			}

			static Sint16
			MS_ADPCM_ProcessNibble(MS_ADPCM_ChannelState *cstate, Sint32 sample1, Sint32 sample2, Uint8 nybble)
			{
			const Sint32 max_audioval = 32767;
			const Sint32 min_audioval = -32768;
			const Uint16 max_deltaval = 65535;
			const Uint16 adaptive[] = {
			230, 230, 230, 230, 307, 409, 512, 614,
			768, 614, 512, 409, 307, 230, 230, 230
			};
			Sint32 new_sample, delta;
			Sint32 new_sample;
			Sint32 errordelta;
			Uint32 delta = cstate->delta;

			new_sample = ((state->iSamp1 * coeff[0]) +
			(state->iSamp2 * coeff[1])) / 256;
			if (nybble & 0x08) {
			new_sample += state->iDelta * (nybble - 0x10);
			} else {
			new_sample += state->iDelta * nybble;
			}
			new_sample = (sample1 * cstate->coeff1 + sample2 * cstate->coeff2) / 256;
			/* The nibble is a signed 4-bit error delta. */
			errordelta = (Sint32)nybble - (nybble >= 0x08 ? 0x10 : 0);
			new_sample += (Sint32)delta * errordelta;
			if (new_sample < min_audioval) {
			new_sample = min_audioval;
			} else if (new_sample > max_audioval) {
			new_sample = max_audioval;
			}
			delta = ((Sint32) state->iDelta * adaptive[nybble]) / 256;
			delta = (delta * adaptive[nybble]) / 256;
			if (delta < 16) {
			delta = 16;
			} else if (delta > max_deltaval) {
			/* This issue is not described in the Standards Update and therefore
			* undefined. It seems sensible to prevent overflows with a limit.
			*/
			delta = max_deltaval;
			}
			state->iDelta = (Uint16) delta;
			state->iSamp2 = state->iSamp1;
			state->iSamp1 = (Sint16) new_sample;
			return (new_sample);

			cstate->delta = (Uint16)delta;
			return (Sint16)new_sample;
			}

			static int
			MS_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
			MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
			{
			struct MS_ADPCM_decodestate *state[2];
			Uint8 freeable, encoded, *decoded;
			Sint32 encoded_len, samplesleft;
			Sint8 nybble;
			Uint8 stereo;
			Sint16 *coeff[2];
			Sint32 new_sample;
			Uint8 coeffindex;
			const Uint32 channels = state->channels;
			Sint32 sample;
			Uint32 c;
			MS_ADPCM_ChannelState cstate = (MS_ADPCM_ChannelState )state->cstate;
			MS_ADPCM_CoeffData ddata = (MS_ADPCM_CoeffData )state->ddata;

			/* Allocate the proper sized output buffer */
			encoded_len = *audio_len;
			encoded = *audio_buf;
			freeable = *audio_buf;
			audio_len = (encoded_len / MS_ADPCM_state.wavefmt.blockalign)
			MS_ADPCM_state.wSamplesPerBlock *
			MS_ADPCM_state.wavefmt.channels * sizeof(Sint16);
			audio_buf = (Uint8 ) SDL_malloc(*audio_len);
			if (*audio_buf == NULL) {
			for (c = 0; c < channels; c++) {
			size_t o = c;

			/* Load the coefficient pair into the channel state. */
			coeffindex = state->block.data[o];
			if (coeffindex > ddata->coeffcount) {
			return SDL_SetError("Invalid MS ADPCM coefficient index in block header");
			}
			cstate[c].coeff1 = ddata->coeff[coeffindex * 2];
			cstate[c].coeff2 = ddata->coeff[coeffindex * 2 + 1];

			/* Initial delta value. */
			o = channels + c * 2;
			cstate[c].delta = state->block.data[o] \| ((Uint16)state->block.data[o + 1] << 8);

			/* Load the samples from the header. Interestingly, the sample later in
			* the output stream comes first.
			*/
			o = channels * 3 + c * 2;
			sample = state->block.data[o] \| ((Sint32)state->block.data[o + 1] << 8);
			if (sample >= 0x8000) {
			sample -= 0x10000;
			}
			state->output.data[state->output.pos + channels] = (Sint16)sample;

			o = channels * 5 + c * 2;
			sample = state->block.data[o] \| ((Sint32)state->block.data[o + 1] << 8);
			if (sample >= 0x8000) {
			sample -= 0x10000;
			}
			state->output.data[state->output.pos] = (Sint16)sample;

			state->output.pos++;
			}

			state->block.pos += state->blockheadersize;

			/* Skip second sample frame that came from the header. */
			state->output.pos += state->channels;

			/* Header provided two sample frames. */
			state->framesleft -= 2;

			return 0;
			}

			/* Decodes the data of the MS ADPCM block. Decoding will stop if a block is too
			* short, returning with none or partially decoded data. The partial data
			* will always contain full sample frames (same sample count for each channel).
			* Incomplete sample frames are discarded.
			*/
			static int
			MS_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
			{
			Uint16 nybble = 0;
			Sint16 sample1, sample2;
			const Uint32 channels = state->channels;
			Uint32 c;
			MS_ADPCM_ChannelState cstate = (MS_ADPCM_ChannelState )state->cstate;

			size_t blockpos = state->block.pos;
			size_t blocksize = state->block.size;

			size_t outpos = state->output.pos;

			Sint64 blockframesleft = state->samplesperblock - 2;
			if (blockframesleft > state->framesleft) {
			blockframesleft = state->framesleft;
			}

			while (blockframesleft > 0) {
			for (c = 0; c < channels; c++) {
			if (nybble & 0x4000) {
			nybble <<= 4;
			} else if (blockpos < blocksize) {
			nybble = state->block.data[blockpos++] \| 0x4000;
			} else {
			/* Out of input data. Drop the incomplete frame and return. */
			state->output.pos = outpos - c;
			return -1;
			}

			/* Load previous samples which may come from the block header. */
			sample1 = state->output.data[outpos - channels];
			sample2 = state->output.data[outpos - channels * 2];

			sample1 = MS_ADPCM_ProcessNibble(cstate + c, sample1, sample2, (nybble >> 4) & 0x0f);
			state->output.data[outpos++] = sample1;
			}

			state->framesleft--;
			blockframesleft--;
			}

			state->output.pos = outpos;

			return 0;
			}

			static int
			MS_ADPCM_Decode(WaveFile file, Uint8 audio_buf, Uint32 audio_len)
			{
			int result;
			size_t bytesleft, outputsize;
			WaveChunk *chunk = &file->chunk;
			ADPCM_DecoderState state;
			MS_ADPCM_ChannelState cstate[2];

			SDL_zero(state);
			SDL_zeroa(cstate);

			if (chunk->size != chunk->length) {
			/* Could not read everything. Recalculate number of sample frames. */
			if (MS_ADPCM_CalculateSampleFrames(file, chunk->size) < 0) {
			return -1;
			}
			}

			/* Nothing to decode, nothing to return. */
			if (file->sampleframes == 0) {
			*audio_buf = NULL;
			*audio_len = 0;
			return 0;
			}

			state.blocksize = file->format.blockalign;
			state.channels = file->format.channels;
			state.blockheadersize = (size_t)state.channels * 7;
			state.samplesperblock = file->format.samplesperblock;
			state.framesize = state.channels * sizeof(Sint16);
			state.ddata = file->decoderdata;
			state.framestotal = file->sampleframes;
			state.framesleft = state.framestotal;

			state.input.data = chunk->data;
			state.input.size = chunk->size;
			state.input.pos = 0;

			/* The output size in bytes. May get modified if data is truncated. */
			outputsize = (size_t)state.framestotal;
			if (SafeMult(&outputsize, state.framesize)) {
			return SDL_OutOfMemory();
			} else if (outputsize > SDL_MAX_UINT32 \|\| state.framestotal > SIZE_MAX) {
			return SDL_SetError("WAVE file too big");
			}

			state.output.pos = 0;
			state.output.size = outputsize / sizeof(Sint16);
			state.output.data = (Sint16 *)SDL_malloc(outputsize);
			if (state.output.data == NULL) {
			return SDL_OutOfMemory();
			}
			decoded = *audio_buf;

			/* Get ready... Go! */
			stereo = (MS_ADPCM_state.wavefmt.channels == 2);
			state[0] = &MS_ADPCM_state.state[0];
			state[1] = &MS_ADPCM_state.state[stereo];
			while (encoded_len >= MS_ADPCM_state.wavefmt.blockalign) {
			/* Grab the initial information for this block */
			state[0]->hPredictor = *encoded++;
			if (stereo) {
			state[1]->hPredictor = *encoded++;
			}
			state[0]->iDelta = ((encoded[1] << 8) \| encoded[0]);
			encoded += sizeof(Sint16);
			if (stereo) {
			state[1]->iDelta = ((encoded[1] << 8) \| encoded[0]);
			encoded += sizeof(Sint16);
			}
			state[0]->iSamp1 = ((encoded[1] << 8) \| encoded[0]);
			encoded += sizeof(Sint16);
			if (stereo) {
			state[1]->iSamp1 = ((encoded[1] << 8) \| encoded[0]);
			encoded += sizeof(Sint16);
			}
			state[0]->iSamp2 = ((encoded[1] << 8) \| encoded[0]);
			encoded += sizeof(Sint16);
			if (stereo) {
			state[1]->iSamp2 = ((encoded[1] << 8) \| encoded[0]);
			encoded += sizeof(Sint16);
			}
			coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor];
			coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor];
			state.cstate = cstate;

			/* Store the two initial samples we start with */
			decoded[0] = state[0]->iSamp2 & 0xFF;
			decoded[1] = state[0]->iSamp2 >> 8;
			decoded += 2;
			if (stereo) {
			decoded[0] = state[1]->iSamp2 & 0xFF;
			decoded[1] = state[1]->iSamp2 >> 8;
			decoded += 2;
			}
			decoded[0] = state[0]->iSamp1 & 0xFF;
			decoded[1] = state[0]->iSamp1 >> 8;
			decoded += 2;
			if (stereo) {
			decoded[0] = state[1]->iSamp1 & 0xFF;
			decoded[1] = state[1]->iSamp1 >> 8;
			decoded += 2;
			/* Decode block by block. A truncated block will stop the decoding. */
			bytesleft = state.input.size - state.input.pos;
			while (state.framesleft > 0 && bytesleft >= state.blockheadersize) {
			state.block.data = state.input.data + state.input.pos;
			state.block.size = bytesleft < state.blocksize ? bytesleft : state.blocksize;
			state.block.pos = 0;

			if (state.output.size - state.output.pos < (Uint64)state.framesleft * state.channels) {
			/* Somehow didn't allocate enough space for the output. */
			SDL_free(state.output.data);
			return SDL_SetError("Unexpected overflow in MS ADPCM decoder");
			}

			/* Decode and store the other samples in this block */
			samplesleft = (MS_ADPCM_state.wSamplesPerBlock - 2) *
			MS_ADPCM_state.wavefmt.channels;
			while (samplesleft > 0) {
			nybble = (*encoded) >> 4;
			new_sample = MS_ADPCM_nibble(state[0], nybble, coeff[0]);
			decoded[0] = new_sample & 0xFF;
			new_sample >>= 8;
			decoded[1] = new_sample & 0xFF;
			decoded += 2;

			nybble = (*encoded) & 0x0F;
			new_sample = MS_ADPCM_nibble(state[1], nybble, coeff[1]);
			decoded[0] = new_sample & 0xFF;
			new_sample >>= 8;
			decoded[1] = new_sample & 0xFF;
			decoded += 2;

			++encoded;
			samplesleft -= 2;
			}
			encoded_len -= MS_ADPCM_state.wavefmt.blockalign;
			}
			SDL_free(freeable);
			return (0);
			/* Initialize decoder with the values from the block header. */
			result = MS_ADPCM_DecodeBlockHeader(&state);
			if (result == -1) {
			SDL_free(state.output.data);
			return -1;
			}

			struct IMA_ADPCM_decodestate
			{
			Sint32 sample;
			Sint8 index;
			};
			static struct IMA_ADPCM_decoder
			{
			WaveFMT wavefmt;
			Uint16 wSamplesPerBlock;
			/* * * */
			struct IMA_ADPCM_decodestate state[2];
			} IMA_ADPCM_state;
			/* Decode the block data. It stores the samples directly in the output. */
			result = MS_ADPCM_DecodeBlockData(&state);
			if (result == -1) {
			/* Unexpected end. Stop decoding and return partial data if necessary. */
			if (file->trunchint == TruncVeryStrict \|\| file->trunchint == TruncStrict) {
			SDL_free(state.output.data);
			return SDL_SetError("Truncated data chunk");
			} else if (file->trunchint != TruncDropFrame) {
			state.output.pos -= state.output.pos % (state.samplesperblock * state.channels);
			}
			outputsize = state.output.pos * sizeof(Sint16); /* Can't overflow, is always smaller. */
			break;
			}

			state.input.pos += state.block.size;
			bytesleft = state.input.size - state.input.pos;
			}

			audio_buf = (Uint8 )state.output.data;
			*audio_len = (Uint32)outputsize;

			return 0;
			}

			static int
			InitIMA_ADPCM(WaveFMT * format)
			IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
			{
			Uint8 *rogue_feel;
			WaveFormat *format = &file->format;
			const size_t blockheadersize = (size_t)format->channels * 4;
			const size_t subblockframesize = (size_t)format->channels * 4;
			const size_t availableblocks = datalength / format->blockalign;
			const size_t trailingdata = datalength % format->blockalign;

			/* Set the rogue pointer to the IMA_ADPCM specific data */
			IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
			IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
			IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
			IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
			IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
			IMA_ADPCM_state.wavefmt.bitspersample =
			SDL_SwapLE16(format->bitspersample);
			rogue_feel = (Uint8 ) format + sizeof(format);
			if (sizeof(*format) == 16) {
			/* const Uint16 extra_info = ((rogue_feel[1] << 8) \| rogue_feel[0]); */
			rogue_feel += sizeof(Uint16);
			if (file->trunchint == TruncVeryStrict \|\| file->trunchint == TruncStrict) {
			/* The size of the data chunk must be a multiple of the block size. */
			if (datalength < blockheadersize \|\| trailingdata > 0) {
			return SDL_SetError("Truncated IMA ADPCM block");
			}
			IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1] << 8) \| rogue_feel[0]);
			return (0);
			}

			static Sint32
			IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state, Uint8 nybble)
			/* Calculate number of sample frames that will be decoded. */
			file->sampleframes = (Uint64)availableblocks * format->samplesperblock;
			if (trailingdata > 0) {
			/* The last block is truncated. Check if we can get any samples out of it. */
			if (file->trunchint == TruncDropFrame && trailingdata > blockheadersize - 2) {
			/* The sample frame in the header of the truncated block is present.
			* Drop incomplete sample frames.
			*/
			size_t trailingsamples = 1;

			if (trailingdata > blockheadersize) {
			/* More data following after the header. */
			const size_t trailingblockdata = trailingdata - blockheadersize;
			const size_t trailingsubblockdata = trailingblockdata % subblockframesize;
			trailingsamples += (trailingblockdata / subblockframesize) * 8;
			/* Due to the interleaved sub-blocks, the last 4 bytes determine
			* how many samples of the truncated sub-block are lost.
			*/
			if (trailingsubblockdata > subblockframesize - 4) {
			trailingsamples += (trailingsubblockdata % 4) * 2;
			}
			}

			if (trailingsamples > format->samplesperblock) {
			trailingsamples = format->samplesperblock;
			}
			file->sampleframes += trailingsamples;
			}
			}

			file->sampleframes = WaveAdjustToFactValue(file, file->sampleframes);
			if (file->sampleframes < 0) {
			return -1;
			}

			return 0;
			}

			static int
			IMA_ADPCM_Init(WaveFile *file, size_t datalength)
			{
			const Sint32 max_audioval = ((1 << (16 - 1)) - 1);
			const Sint32 min_audioval = -(1 << (16 - 1));
			const int index_table[16] = {
			WaveFormat *format = &file->format;
			WaveChunk *chunk = &file->chunk;
			const size_t blockheadersize = (size_t)format->channels * 4;
			const size_t blockdatasize = (size_t)format->blockalign - blockheadersize;
			const size_t blockframebitsize = (size_t)format->bitspersample * format->channels;
			const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize;

			/* Sanity checks. */

			/* IMA ADPCM can also have 3-bit samples, but it's not supported by SDL at this time. */
			if (format->bitspersample == 3) {
			return SDL_SetError("3-bit IMA ADPCM currently not supported");
			} else if (format->bitspersample != 4) {
			return SDL_SetError("Invalid IMA ADPCM bits per sample of %u", (unsigned int)format->bitspersample);
			}

			/* The block size is required to be a multiple of 4 and it must be able to
			* hold a block header.
			*/
			if (format->blockalign < blockheadersize \|\| format->blockalign % 4) {
			return SDL_SetError("Invalid IMA ADPCM block size (nBlockAlign)");
			}

			if (format->formattag == EXTENSIBLE_CODE) {
			/* There's no specification for this, but it's basically the same
			* format because the extensible header has wSampePerBlocks too.
			*/
			} else {
			/* The Standards Update says there 'should' be 2 bytes for wSamplesPerBlock. */
			if (chunk->size >= 20 && format->extsize >= 2) {
			format->samplesperblock = chunk->data[18] \| ((Uint16)chunk->data[19] << 8);
			}
			}

			if (format->samplesperblock == 0) {
			/* Field zero? No problem. We just assume the encoder packed the block.
			* The specification calculates it this way:
			*
			* x = Block size (in bits) minus header size (in bits)
			* y = Bit depth multiplied by channel count
			* z = Number of samples per channel in header
			* wSamplesPerBlock = x / y + z
			*/
			format->samplesperblock = (Uint32)blockdatasamples + 1;
			}

			/* nBlockAlign can be in conflict with wSamplesPerBlock. For example, if
			* the number of samples doesn't fit into the block. The Standards Update
			* also describes wSamplesPerBlock with a formula that makes it necessary
			* to always fill the block with the maximum amount of samples, but this is
			* not enforced here as there are no compatibility issues.
			*/
			if (blockdatasamples < format->samplesperblock - 1) {
			return SDL_SetError("Invalid number of samples per IMA ADPCM block (wSamplesPerBlock)");
			}

			if (IMA_ADPCM_CalculateSampleFrames(file, datalength) < 0) {
			return -1;
			}

			return 0;
			}

			static Sint16
			IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 nybble)
			{
			const Sint32 max_audioval = 32767;
			const Sint32 min_audioval = -32768;
			const Sint8 index_table_4b[16] = {
			-1, -1, -1, -1,
			2, 4, 6, 8,
			-1, -1, -1, -1,
			2, 4, 6, 8
			};
			const Sint32 step_table[89] = {
			const Uint16 step_table[89] = {
			7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31,
			34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130,
			143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408,
			@@ -271,398 +879,1269 @@
			9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,
			22385, 24623, 27086, 29794, 32767
			};
			Sint32 delta, step;
			Uint32 step;
			Sint32 sample, delta;
			Sint8 index = *cindex;

			/* Compute difference and new sample value */
			if (state->index > 88) {
			state->index = 88;
			} else if (state->index < 0) {
			state->index = 0;
			/* Clamp index into valid range. */
			if (index > 88) {
			index = 88;
			} else if (index < 0) {
			index = 0;
			}

			/* explicit cast to avoid gcc warning about using 'char' as array index */
			step = step_table[(int)state->index];
			step = step_table[(size_t)index];

			/* Update index value */
			*cindex = index + index_table_4b[nybble];

			/* This calculation uses shifts and additions because multiplications were
			* much slower back then. Sadly, this can't just be replaced with an actual
			* multiplication now as the old algorithm drops some bits. The closest
			* approximation I could find is something like this:
			* (nybble & 0x8 ? -1 : 1) * ((nybble & 0x7) * step / 4 + step / 8)
			*/
			delta = step >> 3;
			if (nybble & 0x04)
			delta += step;
			if (nybble & 0x02)
			delta += (step >> 1);
			delta += step >> 1;
			if (nybble & 0x01)
			delta += (step >> 2);
			delta += step >> 2;
			if (nybble & 0x08)
			delta = -delta;
			state->sample += delta;

			/* Update index value */
			state->index += index_table[nybble];
			sample = lastsample + delta;

			/* Clamp output sample */
			if (state->sample > max_audioval) {
			state->sample = max_audioval;
			} else if (state->sample < min_audioval) {
			state->sample = min_audioval;
			}
			return (state->sample);
			if (sample > max_audioval) {
			sample = max_audioval;
			} else if (sample < min_audioval) {
			sample = min_audioval;
			}

			/* Fill the decode buffer with a channel block of data (8 samples) */
			static void
			Fill_IMA_ADPCM_block(Uint8 * decoded, Uint8 * encoded,
			int channel, int numchannels,
			struct IMA_ADPCM_decodestate *state)
			{
			int i;
			Sint8 nybble;
			Sint32 new_sample;

			decoded += (channel * 2);
			for (i = 0; i < 4; ++i) {
			nybble = (*encoded) & 0x0F;
			new_sample = IMA_ADPCM_nibble(state, nybble);
			decoded[0] = new_sample & 0xFF;
			new_sample >>= 8;
			decoded[1] = new_sample & 0xFF;
			decoded += 2 * numchannels;

			nybble = (*encoded) >> 4;
			new_sample = IMA_ADPCM_nibble(state, nybble);
			decoded[0] = new_sample & 0xFF;
			new_sample >>= 8;
			decoded[1] = new_sample & 0xFF;
			decoded += 2 * numchannels;

			++encoded;
			}
			return (Sint16)sample;
			}

			static int
			IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
			IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
			{
			struct IMA_ADPCM_decodestate *state;
			Uint8 freeable, encoded, *decoded;
			Sint32 encoded_len, samplesleft;
			unsigned int c, channels;
			Sint16 step;
			Uint32 c;
			Uint8 *cstate = state->cstate;

			/* Check to make sure we have enough variables in the state array */
			channels = IMA_ADPCM_state.wavefmt.channels;
			if (channels > SDL_arraysize(IMA_ADPCM_state.state)) {
			SDL_SetError("IMA ADPCM decoder can only handle %u channels",
			(unsigned int)SDL_arraysize(IMA_ADPCM_state.state));
			return (-1);
			}
			state = IMA_ADPCM_state.state;
			for (c = 0; c < state->channels; c++) {
			size_t o = state->block.pos + c * 4;

			/* Allocate the proper sized output buffer */
			encoded_len = *audio_len;
			encoded = *audio_buf;
			freeable = *audio_buf;
			audio_len = (encoded_len / IMA_ADPCM_state.wavefmt.blockalign)
			IMA_ADPCM_state.wSamplesPerBlock *
			IMA_ADPCM_state.wavefmt.channels * sizeof(Sint16);
			audio_buf = (Uint8 ) SDL_malloc(*audio_len);
			if (*audio_buf == NULL) {
			return SDL_OutOfMemory();
			/* Extract the sample from the header. */
			Sint32 sample = state->block.data[o] \| ((Sint32)state->block.data[o + 1] << 8);
			if (sample >= 0x8000) {
			sample -= 0x10000;
			}
			decoded = *audio_buf;
			state->output.data[state->output.pos++] = (Sint16)sample;

			/* Get ready... Go! */
			while (encoded_len >= IMA_ADPCM_state.wavefmt.blockalign) {
			/* Grab the initial information for this block */
			for (c = 0; c < channels; ++c) {
			/* Fill the state information for this block */
			state[c].sample = ((encoded[1] << 8) \| encoded[0]);
			encoded += 2;
			if (state[c].sample & 0x8000) {
			state[c].sample -= 0x10000;
			}
			state[c].index = *encoded++;
			/* Reserved byte in buffer header, should be 0 */
			if (*encoded++ != 0) {
			/* Channel step index. */
			step = (Sint16)state->block.data[o + 2];
			cstate[c] = (Sint8)(step > 0x80 ? step - 0x100 : step);

			/* Reserved byte in block header, should be 0. */
			if (state->block.data[o + 3] != 0) {
			/* Uh oh, corrupt data? Buggy code? */ ;
			}

			/* Store the initial sample we start with */
			decoded[0] = (Uint8) (state[c].sample & 0xFF);
			decoded[1] = (Uint8) (state[c].sample >> 8);
			decoded += 2;
			}

			/* Decode and store the other samples in this block */
			samplesleft = (IMA_ADPCM_state.wSamplesPerBlock - 1) * channels;
			while (samplesleft > 0) {
			for (c = 0; c < channels; ++c) {
			Fill_IMA_ADPCM_block(decoded, encoded,
			c, channels, &state[c]);
			encoded += 4;
			samplesleft -= 8;
			}
			decoded += (channels * 8 * 2);
			}
			encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
			}
			SDL_free(freeable);
			return (0);
			state->block.pos += state->blockheadersize;

			/* Header provided one sample frame. */
			state->framesleft--;

			return 0;
			}

			/* Decodes the data of the IMA ADPCM block. Decoding will stop if a block is too
			* short, returning with none or partially decoded data. The partial data always
			* contains full sample frames (same sample count for each channel).
			* Incomplete sample frames are discarded.
			*/
			static int
			IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
			{
			size_t i;
			int retval = 0;
			const Uint32 channels = state->channels;
			const size_t subblockframesize = channels * 4;
			Uint64 bytesrequired;
			Uint32 c;

			size_t blockpos = state->block.pos;
			size_t blocksize = state->block.size;
			size_t blockleft = blocksize - blockpos;

			size_t outpos = state->output.pos;

			Sint64 blockframesleft = state->samplesperblock - 1;
			if (blockframesleft > state->framesleft) {
			blockframesleft = state->framesleft;
			}

			bytesrequired = (blockframesleft + 7) / 8 * subblockframesize;
			if (blockleft < bytesrequired) {
			/* Data truncated. Calculate how many samples we can get out if it. */
			const size_t guaranteedframes = blockleft / subblockframesize;
			const size_t remainingbytes = blockleft % subblockframesize;
			blockframesleft = guaranteedframes;
			if (remainingbytes > subblockframesize - 4) {
			blockframesleft += (remainingbytes % 4) * 2;
			}
			/* Signal the truncation. */
			retval = -1;
			}

			/* Each channel has their nibbles packed into 32-bit blocks. These blocks
			* are interleaved and make up the data part of the ADPCM block. This loop
			* decodes the samples as they come from the input data and puts them at
			* the appropriate places in the output data.
			*/
			while (blockframesleft > 0) {
			const size_t subblocksamples = blockframesleft < 8 ? (size_t)blockframesleft : 8;

			for (c = 0; c < channels; c++) {
			Uint8 nybble = 0;
			/* Load previous sample which may come from the block header. */
			Sint16 sample = state->output.data[outpos + c - channels];

			for (i = 0; i < subblocksamples; i++) {
			if (i & 1) {
			nybble >>= 4;
			} else {
			nybble = state->block.data[blockpos++];
			}

			sample = IMA_ADPCM_ProcessNibble((Sint8 *)state->cstate + c, sample, nybble & 0x0f);
			state->output.data[outpos + c + i * channels] = sample;
			}
			}

			outpos += channels * subblocksamples;
			state->framesleft -= subblocksamples;
			blockframesleft -= subblocksamples;
			}

			state->block.pos = blockpos;
			state->output.pos = outpos;

			return retval;
			}

			static int
			ConvertSint24ToSint32(Uint8 ** audio_buf, Uint32 * audio_len)
			IMA_ADPCM_Decode(WaveFile file, Uint8 audio_buf, Uint32 audio_len)
			{
			const double DIVBY8388608 = 0.00000011920928955078125;
			const Uint32 original_len = *audio_len;
			const Uint32 samples = original_len / 3;
			const Uint32 expanded_len = samples * sizeof (Uint32);
			Uint8 ptr = (Uint8 ) SDL_realloc(*audio_buf, expanded_len);
			const Uint8 *src;
			Uint32 *dst;
			Uint32 i;
			int result;
			size_t bytesleft, outputsize;
			WaveChunk *chunk = &file->chunk;
			ADPCM_DecoderState state;
			Sint8 *cstate;

			if (!ptr) {
			if (chunk->size != chunk->length) {
			/* Could not read everything. Recalculate number of sample frames. */
			if (IMA_ADPCM_CalculateSampleFrames(file, chunk->size) < 0) {
			return -1;
			}
			}

			/* Nothing to decode, nothing to return. */
			if (file->sampleframes == 0) {
			*audio_buf = NULL;
			*audio_len = 0;
			return 0;
			}

			SDL_zero(state);
			state.channels = file->format.channels;
			state.blocksize = file->format.blockalign;
			state.blockheadersize = (size_t)state.channels * 4;
			state.samplesperblock = file->format.samplesperblock;
			state.framesize = state.channels * sizeof(Sint16);
			state.framestotal = file->sampleframes;
			state.framesleft = state.framestotal;

			state.input.data = chunk->data;
			state.input.size = chunk->size;
			state.input.pos = 0;

			/* The output size in bytes. May get modified if data is truncated. */
			outputsize = (size_t)state.framestotal;
			if (SafeMult(&outputsize, state.framesize)) {
			return SDL_OutOfMemory();
			} else if (outputsize > SDL_MAX_UINT32 \|\| state.framestotal > SIZE_MAX) {
			return SDL_SetError("WAVE file too big");
			}

			state.output.pos = 0;
			state.output.size = outputsize / sizeof(Sint16);
			state.output.data = (Sint16 *)SDL_malloc(outputsize);
			if (state.output.data == NULL) {
			return SDL_OutOfMemory();
			}

			*audio_buf = ptr;
			*audio_len = expanded_len;
			cstate = (Sint8 *)SDL_calloc(state.channels, sizeof(Sint8));
			if (cstate == NULL) {
			SDL_free(state.output.data);
			return SDL_OutOfMemory();
			}
			state.cstate = cstate;

			/* work from end to start, since we're expanding in-place. */
			src = (ptr + original_len) - 3;
			dst = ((Uint32 *) (ptr + expanded_len)) - 1;
			for (i = 0; i < samples; i++) {
			/* There's probably a faster way to do all this. */
			const Sint32 converted = ((Sint32) ( (((Uint32) src[2]) << 24) \|
			(((Uint32) src[1]) << 16) \|
			(((Uint32) src[0]) << 8) )) >> 8;
			const double scaled = (((double) converted) * DIVBY8388608);
			src -= 3;
			(dst--) = (Sint32) (scaled 2147483647.0);
			/* Decode block by block. A truncated block will stop the decoding. */
			bytesleft = state.input.size - state.input.pos;
			while (state.framesleft > 0 && bytesleft >= state.blockheadersize) {
			state.block.data = state.input.data + state.input.pos;
			state.block.size = bytesleft < state.blocksize ? bytesleft : state.blocksize;
			state.block.pos = 0;

			if (state.output.size - state.output.pos < (Uint64)state.framesleft * state.channels) {
			/* Somehow didn't allocate enough space for the output. */
			SDL_free(state.output.data);
			SDL_free(cstate);
			return SDL_SetError("Unexpected overflow in IMA ADPCM decoder");
			}

			/* Initialize decoder with the values from the block header. */
			result = IMA_ADPCM_DecodeBlockHeader(&state);
			if (result == 0) {
			/* Decode the block data. It stores the samples directly in the output. */
			result = IMA_ADPCM_DecodeBlockData(&state);
			}

			if (result == -1) {
			/* Unexpected end. Stop decoding and return partial data if necessary. */
			if (file->trunchint == TruncVeryStrict \|\| file->trunchint == TruncStrict) {
			SDL_free(state.output.data);
			SDL_free(cstate);
			return SDL_SetError("Truncated data chunk");
			} else if (file->trunchint != TruncDropFrame) {
			state.output.pos -= state.output.pos % (state.samplesperblock * state.channels);
			}
			outputsize = state.output.pos * sizeof(Sint16); /* Can't overflow, is always smaller. */
			break;
			}

			state.input.pos += state.block.size;
			bytesleft = state.input.size - state.input.pos;
			}

			audio_buf = (Uint8 )state.output.data;
			*audio_len = (Uint32)outputsize;

			SDL_free(cstate);

			return 0;
			}

			static int
			LAW_Init(WaveFile *file, size_t datalength)
			{
			WaveFormat *format = &file->format;

			/* Standards Update requires this to be 8. */
			if (format->bitspersample != 8) {
			return SDL_SetError("Invalid companded bits per sample of %u", (unsigned int)format->bitspersample);
			}

			/* Not going to bother with weird padding. */
			if (format->blockalign != format->channels) {
			return SDL_SetError("Unsupported block alignment");
			}

			if ((file->trunchint == TruncVeryStrict \|\| file->trunchint == TruncStrict)) {
			if (format->blockalign > 1 && datalength % format->blockalign) {
			return SDL_SetError("Truncated data chunk in WAVE file");
			}
			}

			file->sampleframes = WaveAdjustToFactValue(file, datalength / format->blockalign);
			if (file->sampleframes < 0) {
			return -1;
			}

			return 0;
			}


			/* GUIDs that are used by WAVE_FORMAT_EXTENSIBLE */
			static const Uint8 extensible_pcm_guid[16] = { 1, 0, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 };
			static const Uint8 extensible_ieee_guid[16] = { 3, 0, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 };

			SDL_AudioSpec *
			SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
			SDL_AudioSpec * spec, Uint8 ** audio_buf, Uint32 * audio_len)
			static int
			LAW_Decode(WaveFile file, Uint8 audio_buf, Uint32 audio_len)
			{
			int was_error;
			Chunk chunk;
			int lenread;
			int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
			int samplesize;
			#ifdef SDL_WAVE_LAW_LUT
			const Sint16 alaw_lut[256] = {
			-5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, -2752,
			-2624, -3008, -2880, -2240, -2112, -2496, -2368, -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, -22016,
			-20992, -24064, -23040, -17920, -16896, -19968, -18944, -30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136, -11008,
			-10496, -12032, -11520, -8960, -8448, -9984, -9472, -15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568, -344,
			-328, -376, -360, -280, -264, -312, -296, -472, -456, -504, -488, -408, -392, -440, -424, -88,
			-72, -120, -104, -24, -8, -56, -40, -216, -200, -248, -232, -152, -136, -184, -168, -1376,
			-1312, -1504, -1440, -1120, -1056, -1248, -1184, -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, -688,
			-656, -752, -720, -560, -528, -624, -592, -944, -912, -1008, -976, -816, -784, -880, -848, 5504,
			5248, 6016, 5760, 4480, 4224, 4992, 4736, 7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784, 2752,
			2624, 3008, 2880, 2240, 2112, 2496, 2368, 3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392, 22016,
			20992, 24064, 23040, 17920, 16896, 19968, 18944, 30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136, 11008,
			10496, 12032, 11520, 8960, 8448, 9984, 9472, 15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568, 344,
			328, 376, 360, 280, 264, 312, 296, 472, 456, 504, 488, 408, 392, 440, 424, 88,
			72, 120, 104, 24, 8, 56, 40, 216, 200, 248, 232, 152, 136, 184, 168, 1376,
			1312, 1504, 1440, 1120, 1056, 1248, 1184, 1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696, 688,
			656, 752, 720, 560, 528, 624, 592, 944, 912, 1008, 976, 816, 784, 880, 848
			};
			const Sint16 mulaw_lut[256] = {
			-32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, -15996,
			-15484, -14972, -14460, -13948, -13436, -12924, -12412, -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, -7932,
			-7676, -7420, -7164, -6908, -6652, -6396, -6140, -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, -3900,
			-3772, -3644, -3516, -3388, -3260, -3132, -3004, -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, -1884,
			-1820, -1756, -1692, -1628, -1564, -1500, -1436, -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, -876,
			-844, -812, -780, -748, -716, -684, -652, -620, -588, -556, -524, -492, -460, -428, -396, -372,
			-356, -340, -324, -308, -292, -276, -260, -244, -228, -212, -196, -180, -164, -148, -132, -120,
			-112, -104, -96, -88, -80, -72, -64, -56, -48, -40, -32, -24, -16, -8, 0, 32124,
			31100, 30076, 29052, 28028, 27004, 25980, 24956, 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, 15996,
			15484, 14972, 14460, 13948, 13436, 12924, 12412, 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, 7932,
			7676, 7420, 7164, 6908, 6652, 6396, 6140, 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, 3900,
			3772, 3644, 3516, 3388, 3260, 3132, 3004, 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, 1884,
			1820, 1756, 1692, 1628, 1564, 1500, 1436, 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, 876,
			844, 812, 780, 748, 716, 684, 652, 620, 588, 556, 524, 492, 460, 428, 396, 372,
			356, 340, 324, 308, 292, 276, 260, 244, 228, 212, 196, 180, 164, 148, 132, 120,
			112, 104, 96, 88, 80, 72, 64, 56, 48, 40, 32, 24, 16, 8, 0
			};
			#endif

			/* WAV magic header */
			Uint32 RIFFchunk;
			Uint32 wavelen = 0;
			Uint32 WAVEmagic;
			Uint32 headerDiff = 0;
			WaveFormat *format = &file->format;
			WaveChunk *chunk = &file->chunk;
			size_t i, sample_count, expanded_len;
			Uint8 *src;
			Sint16 *dst;

			/* FMT chunk */
			WaveFMT *format = NULL;
			WaveExtensibleFMT *ext = NULL;
			if (chunk->length != chunk->size) {
			file->sampleframes = WaveAdjustToFactValue(file, chunk->size / format->blockalign);
			if (file->sampleframes < 0) {
			return -1;
			}
			}

			SDL_zero(chunk);
			/* Nothing to decode, nothing to return. */
			if (file->sampleframes == 0) {
			*audio_buf = NULL;
			*audio_len = 0;
			return 0;
			}

			/* Make sure we are passed a valid data source */
			was_error = 0;
			sample_count = (size_t)file->sampleframes;
			if (SafeMult(&sample_count, format->channels)) {
			return SDL_OutOfMemory();
			}

			expanded_len = sample_count;
			if (SafeMult(&expanded_len, sizeof(Sint16))) {
			return SDL_OutOfMemory();
			} else if (expanded_len > SDL_MAX_UINT32 \|\| file->sampleframes > SIZE_MAX) {
			return SDL_SetError("WAVE file too big");
			}

			/* 1 to avoid allocating zero bytes, to keep static analysis happy. */
			src = (Uint8 *)SDL_realloc(chunk->data, expanded_len ? expanded_len : 1);
			if (src == NULL) {
			was_error = 1;
			goto done;
			return SDL_OutOfMemory();
			}
			chunk->data = NULL;
			chunk->size = 0;

			/* Check the magic header */
			RIFFchunk = SDL_ReadLE32(src);
			wavelen = SDL_ReadLE32(src);
			if (wavelen == WAVE) { /* The RIFFchunk has already been read */
			WAVEmagic = wavelen;
			wavelen = RIFFchunk;
			RIFFchunk = RIFF;
			} else {
			WAVEmagic = SDL_ReadLE32(src);
			}
			if ((RIFFchunk != RIFF) \|\| (WAVEmagic != WAVE)) {
			SDL_SetError("Unrecognized file type (not WAVE)");
			was_error = 1;
			goto done;
			}
			headerDiff += sizeof(Uint32); /* for WAVE */
			dst = (Sint16 *)src;

			/* Read the audio data format chunk */
			chunk.data = NULL;
			do {
			SDL_free(chunk.data);
			chunk.data = NULL;
			lenread = ReadChunk(src, &chunk);
			if (lenread < 0) {
			was_error = 1;
			goto done;
			/* Work backwards, since we're expanding in-place. SDL_AudioSpec.format will
			* inform the caller about the byte order.
			*/
			i = sample_count;
			switch (file->format.encoding) {
			#ifdef SDL_WAVE_LAW_LUT
			case ALAW_CODE:
			while (i--) {
			dst[i] = alaw_lut[src[i]];
			}
			/* 2 Uint32's for chunk header+len, plus the lenread */
			headerDiff += lenread + 2 * sizeof(Uint32);
			} while ((chunk.magic == FACT) \|\| (chunk.magic == LIST) \|\| (chunk.magic == BEXT) \|\| (chunk.magic == JUNK));

			/* Decode the audio data format */
			format = (WaveFMT *) chunk.data;
			if (chunk.magic != FMT) {
			SDL_SetError("Complex WAVE files not supported");
			was_error = 1;
			goto done;
			}
			IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
			switch (SDL_SwapLE16(format->encoding)) {
			case PCM_CODE:
			/* We can understand this */
			break;
			case MULAW_CODE:
			while (i--) {
			dst[i] = mulaw_lut[src[i]];
			}
			break;
			#else
			case ALAW_CODE:
			while (i--) {
			Uint8 nibble = src[i];
			Uint8 exponent = (nibble & 0x7f) ^ 0x55;
			Sint16 mantissa = exponent & 0xf;

			exponent >>= 4;
			if (exponent > 0) {
			mantissa \|= 0x10;
			}
			mantissa = (mantissa << 4) \| 0x8;
			if (exponent > 1) {
			mantissa <<= exponent - 1;
			}

			dst[i] = nibble & 0x80 ? mantissa : -mantissa;
			}
			break;
			case MULAW_CODE:
			while (i--) {
			Uint8 nibble = ~src[i];
			Sint16 mantissa = nibble & 0xf;
			Uint8 exponent = (nibble >> 4) & 0x7;
			Sint16 step = 4 << (exponent + 1);

			mantissa = (0x80 << exponent) + step * mantissa + step / 2 - 132;

			dst[i] = nibble & 0x80 ? -mantissa : mantissa;
			}
			break;
			#endif
			default:
			SDL_free(src);
			return SDL_SetError("Unknown companded encoding");
			}

			*audio_buf = src;
			*audio_len = (Uint32)expanded_len;

			return 0;
			}

			static int
			PCM_Init(WaveFile *file, size_t datalength)
			{
			WaveFormat *format = &file->format;

			if (format->encoding == PCM_CODE) {
			switch (format->bitspersample) {
			case 8:
			case 16:
			case 24:
			case 32:
			/* These are supported. */
			break;
			default:
			return SDL_SetError("%u-bit PCM format not supported", (unsigned int)format->bitspersample);
			}
			} else if (format->encoding == IEEE_FLOAT_CODE) {
			if (format->bitspersample != 32) {
			return SDL_SetError("%u-bit IEEE floating-point format not supported", (unsigned int)format->bitspersample);
			}
			}

			/* It wouldn't be that hard to support more exotic block sizes, but
			* the most common formats should do for now.
			*/
			if (format->blockalign * 8 != format->channels * format->bitspersample) {
			return SDL_SetError("Unsupported block alignment");
			}

			if ((file->trunchint == TruncVeryStrict \|\| file->trunchint == TruncStrict)) {
			if (format->blockalign > 1 && datalength % format->blockalign) {
			return SDL_SetError("Truncated data chunk in WAVE file");
			}
			}

			file->sampleframes = WaveAdjustToFactValue(file, datalength / format->blockalign);
			if (file->sampleframes < 0) {
			return -1;
			}

			return 0;
			}

			static int
			PCM_ConvertSint24ToSint32(WaveFile file, Uint8 audio_buf, Uint32 audio_len)
			{
			WaveFormat *format = &file->format;
			WaveChunk *chunk = &file->chunk;
			size_t i, expanded_len, sample_count;
			Uint8 *ptr;

			sample_count = (size_t)file->sampleframes;
			if (SafeMult(&sample_count, format->channels)) {
			return SDL_OutOfMemory();
			}

			expanded_len = sample_count;
			if (SafeMult(&expanded_len, sizeof(Sint32))) {
			return SDL_OutOfMemory();
			} else if (expanded_len > SDL_MAX_UINT32 \|\| file->sampleframes > SIZE_MAX) {
			return SDL_SetError("WAVE file too big");
			}

			/* 1 to avoid allocating zero bytes, to keep static analysis happy. */
			ptr = (Uint8 *)SDL_realloc(chunk->data, expanded_len ? expanded_len : 1);
			if (ptr == NULL) {
			return SDL_OutOfMemory();
			}

			/* This pointer is now invalid. */
			chunk->data = NULL;
			chunk->size = 0;

			*audio_buf = ptr;
			*audio_len = (Uint32)expanded_len;

			/* work from end to start, since we're expanding in-place. */
			for (i = sample_count; i > 0; i--) {
			const size_t o = i - 1;
			uint8_t b[4];

			b[0] = 0;
			b[1] = ptr[o * 3];
			b[2] = ptr[o * 3 + 1];
			b[3] = ptr[o * 3 + 2];

			ptr[o * 4 + 0] = b[0];
			ptr[o * 4 + 1] = b[1];
			ptr[o * 4 + 2] = b[2];
			ptr[o * 4 + 3] = b[3];
			}

			return 0;
			}

			static int
			PCM_Decode(WaveFile file, Uint8 audio_buf, Uint32 audio_len)
			{
			WaveFormat *format = &file->format;
			WaveChunk *chunk = &file->chunk;
			size_t outputsize;

			if (chunk->length != chunk->size) {
			file->sampleframes = WaveAdjustToFactValue(file, chunk->size / format->blockalign);
			if (file->sampleframes < 0) {
			return -1;
			}
			}

			/* Nothing to decode, nothing to return. */
			if (file->sampleframes == 0) {
			*audio_buf = NULL;
			*audio_len = 0;
			return 0;
			}

			/* 24-bit samples get shifted to 32 bits. */
			if (format->encoding == PCM_CODE && format->bitspersample == 24) {
			return PCM_ConvertSint24ToSint32(file, audio_buf, audio_len);
			}

			outputsize = (size_t)file->sampleframes;
			if (SafeMult(&outputsize, format->blockalign)) {
			return SDL_OutOfMemory();
			} else if (outputsize > SDL_MAX_UINT32 \|\| file->sampleframes > SIZE_MAX) {
			return SDL_SetError("WAVE file too big");
			}

			*audio_buf = chunk->data;
			*audio_len = (Uint32)outputsize;

			/* This pointer is going to be returned to the caller. Prevent free in cleanup. */
			chunk->data = NULL;
			chunk->size = 0;

			return 0;
			}

			static WaveRiffSizeHint
			WaveGetRiffSizeHint()
			{
			const char *hint = SDL_GetHint(SDL_HINT_WAVE_RIFF_CHUNK_SIZE);

			if (hint != NULL) {
			if (SDL_strcmp(hint, "force") == 0) {
			return RiffSizeForce;
			} else if (SDL_strcmp(hint, "ignore") == 0) {
			return RiffSizeIgnore;
			} else if (SDL_strcmp(hint, "ignorezero") == 0) {
			return RiffSizeIgnoreZero;
			} else if (SDL_strcmp(hint, "maximum") == 0) {
			return RiffSizeMaximum;
			}
			}

			return RiffSizeNoHint;
			}

			static WaveTruncationHint
			WaveGetTruncationHint()
			{
			const char *hint = SDL_GetHint(SDL_HINT_WAVE_TRUNCATION);

			if (hint != NULL) {
			if (SDL_strcmp(hint, "verystrict") == 0) {
			return TruncVeryStrict;
			} else if (SDL_strcmp(hint, "strict") == 0) {
			return TruncStrict;
			} else if (SDL_strcmp(hint, "dropframe") == 0) {
			return TruncDropFrame;
			} else if (SDL_strcmp(hint, "dropblock") == 0) {
			return TruncDropBlock;
			}
			}

			return TruncNoHint;
			}

			static WaveFactChunkHint
			WaveGetFactChunkHint()
			{
			const char *hint = SDL_GetHint(SDL_HINT_WAVE_FACT_CHUNK);

			if (hint != NULL) {
			if (SDL_strcmp(hint, "truncate") == 0) {
			return FactTruncate;
			} else if (SDL_strcmp(hint, "strict") == 0) {
			return FactStrict;
			} else if (SDL_strcmp(hint, "ignorezero") == 0) {
			return FactIgnoreZero;
			} else if (SDL_strcmp(hint, "ignore") == 0) {
			return FactIgnore;
			}
			}

			return FactNoHint;
			}

			static void
			WaveFreeChunkData(WaveChunk *chunk)
			{
			if (chunk->data != NULL) {
			SDL_free(chunk->data);
			chunk->data = NULL;
			}
			chunk->size = 0;
			}

			static int
			WaveNextChunk(SDL_RWops src, WaveChunk chunk)
			{
			Uint32 chunkheader[2];
			Sint64 nextposition = chunk->position + chunk->length;

			/* Data is no longer valid after this function returns. */
			WaveFreeChunkData(chunk);

			/* Error on overflows. */
			if (SDL_MAX_SINT64 - chunk->length < chunk->position \|\| SDL_MAX_SINT64 - 8 < nextposition) {
			return -1;
			}

			/* RIFF chunks have a 2-byte alignment. Skip padding byte. */
			if (chunk->length & 1) {
			nextposition++;
			}

			if (SDL_RWseek(src, nextposition, RW_SEEK_SET) != nextposition) {
			/* Not sure how we ended up here. Just abort. */
			return -2;
			} else if (SDL_RWread(src, chunkheader, 4, 2) != 2) {
			return -1;
			}

			chunk->fourcc = SDL_SwapLE32(chunkheader[0]);
			chunk->length = SDL_SwapLE32(chunkheader[1]);
			chunk->position = nextposition + 8;

			return 0;
			}

			static int
			WaveReadPartialChunkData(SDL_RWops src, WaveChunk chunk, size_t length)
			{
			WaveFreeChunkData(chunk);

			if (length > chunk->length) {
			length = chunk->length;
			}

			if (length > 0) {
			chunk->data = SDL_malloc(length);
			if (chunk->data == NULL) {
			return SDL_OutOfMemory();
			}

			if (SDL_RWseek(src, chunk->position, RW_SEEK_SET) != chunk->position) {
			/* Not sure how we ended up here. Just abort. */
			return -2;
			}

			chunk->size = SDL_RWread(src, chunk->data, 1, length);
			if (chunk->size != length) {
			/* Expected to be handled by the caller. */
			}
			}

			return 0;
			}

			static int
			WaveReadChunkData(SDL_RWops src, WaveChunk chunk)
			{
			return WaveReadPartialChunkData(src, chunk, chunk->length);
			}

			typedef struct WaveExtensibleGUID {
			Uint16 encoding;
			Uint8 guid[16];
			} WaveExtensibleGUID;

			/* Some of the GUIDs that are used by WAVEFORMATEXTENSIBLE. */
			#define WAVE_FORMATTAG_GUID(tag) {(tag) & 0xff, (tag) >> 8, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113}
			static WaveExtensibleGUID extensible_guids[] = {
			{PCM_CODE, WAVE_FORMATTAG_GUID(PCM_CODE)},
			{MS_ADPCM_CODE, WAVE_FORMATTAG_GUID(MS_ADPCM_CODE)},
			{IEEE_FLOAT_CODE, WAVE_FORMATTAG_GUID(IEEE_FLOAT_CODE)},
			{ALAW_CODE, WAVE_FORMATTAG_GUID(ALAW_CODE)},
			{MULAW_CODE, WAVE_FORMATTAG_GUID(MULAW_CODE)},
			{IMA_ADPCM_CODE, WAVE_FORMATTAG_GUID(IMA_ADPCM_CODE)}
			};

			static Uint16
			WaveGetFormatGUIDEncoding(WaveFormat *format)
			{
			size_t i;
			for (i = 0; i < SDL_arraysize(extensible_guids); i++) {
			if (SDL_memcmp(format->subformat, extensible_guids[i].guid, 16) == 0) {
			return extensible_guids[i].encoding;
			}
			}
			return UNKNOWN_CODE;
			}

			static int
			WaveReadFormat(WaveFile *file)
			{
			WaveChunk *chunk = &file->chunk;
			WaveFormat *format = &file->format;
			SDL_RWops *fmtsrc;
			size_t fmtlen = chunk->size;

			if (fmtlen > SDL_MAX_SINT32) {
			/* Limit given by SDL_RWFromConstMem. */
			return SDL_SetError("Data of WAVE fmt chunk too big");
			}
			fmtsrc = SDL_RWFromConstMem(chunk->data, (int)chunk->size);
			if (fmtsrc == NULL) {
			return SDL_OutOfMemory();
			}

			format->formattag = SDL_ReadLE16(fmtsrc);
			format->encoding = format->formattag;
			format->channels = SDL_ReadLE16(fmtsrc);
			format->frequency = SDL_ReadLE32(fmtsrc);
			format->byterate = SDL_ReadLE32(fmtsrc);
			format->blockalign = SDL_ReadLE16(fmtsrc);

			/* This is PCM specific in the first version of the specification. */
			if (fmtlen >= 16) {
			format->bitspersample = SDL_ReadLE16(fmtsrc);
			} else if (format->encoding == PCM_CODE) {
			SDL_RWclose(fmtsrc);
			return SDL_SetError("Missing wBitsPerSample field in WAVE fmt chunk");
			}

			/* The earlier versions also don't have this field. */
			if (fmtlen >= 18) {
			format->extsize = SDL_ReadLE16(fmtsrc);
			}

			if (format->formattag == EXTENSIBLE_CODE) {
			/* note that this ignores channel masks, smaller valid bit counts
			* inside a larger container, and most subtypes. This is just enough
			* to get things that didn't really _need_ WAVE_FORMAT_EXTENSIBLE
			* to be useful working when they use this format flag.
			*/

			/* Extensible header must be at least 22 bytes. */
			if (fmtlen < 40 \|\| format->extsize < 22) {
			SDL_RWclose(fmtsrc);
			return SDL_SetError("Extensible WAVE header too small");
			}

			format->validsamplebits = SDL_ReadLE16(fmtsrc);
			format->samplesperblock = format->validsamplebits;
			format->channelmask = SDL_ReadLE32(fmtsrc);
			SDL_RWread(fmtsrc, format->subformat, 1, 16);
			format->encoding = WaveGetFormatGUIDEncoding(format);
			}

			SDL_RWclose(fmtsrc);

			return 0;
			}

			static int
			WaveCheckFormat(WaveFile *file, size_t datalength)
			{
			WaveFormat *format = &file->format;

			/* Check for some obvious issues. */

			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");
			}

			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);
			}

			/* Reject invalid fact chunks in strict mode. */
			if (file->facthint == FactStrict && file->fact.status == -1) {
			return SDL_SetError("Invalid fact chunk in WAVE file");
			}

			/* Check for issues common to all encodings. Some unsupported formats set
			* the bits per sample to zero. These fall through to the 'unsupported
			* format' error.
			*/
			switch (format->encoding) {
			case IEEE_FLOAT_CODE:
			IEEE_float_encoded = 1;
			/* We can understand this */
			case ALAW_CODE:
			case MULAW_CODE:
			case MS_ADPCM_CODE:
			case IMA_ADPCM_CODE:
			/* These formats require a fact chunk. */
			if (file->facthint == FactStrict && file->fact.status <= 0) {
			return SDL_SetError("Missing fact chunk in WAVE file");
			}
			/* fallthrough */
			case PCM_CODE:
			/* All supported formats require a non-zero bit depth. */
			if (file->chunk.size < 16) {
			return SDL_SetError("Missing wBitsPerSample field in WAVE fmt chunk");
			} else if (format->bitspersample == 0) {
			return SDL_SetError("Invalid bits per sample");
			}

			/* All supported formats must have a proper block size. */
			if (format->blockalign == 0) {
			return SDL_SetError("Invalid block alignment");
			}

			/* If the fact chunk is valid and the appropriate hint is set, the
			* decoders will use the number of sample frames from the fact chunk.
			*/
			if (file->fact.status == 1) {
			WaveFactChunkHint hint = file->facthint;
			Uint32 samples = file->fact.samplelength;
			if (hint == FactTruncate \|\| hint == FactStrict \|\| (hint == FactIgnoreZero && samples > 0)) {
			file->fact.status = 2;
			}
			}
			}

			/* Check the format for encoding specific issues and initialize decoders. */
			switch (format->encoding) {
			case PCM_CODE:
			case IEEE_FLOAT_CODE:
			if (PCM_Init(file, datalength) < 0) {
			return -1;
			}
			break;
			case ALAW_CODE:
			case MULAW_CODE:
			if (LAW_Init(file, datalength) < 0) {
			return -1;
			}
			break;
			case MS_ADPCM_CODE:
			/* Try to understand this */
			if (InitMS_ADPCM(format) < 0) {
			was_error = 1;
			goto done;
			if (MS_ADPCM_Init(file, datalength) < 0) {
			return -1;
			}
			MS_ADPCM_encoded = 1;
			break;
			case IMA_ADPCM_CODE:
			/* Try to understand this */
			if (InitIMA_ADPCM(format) < 0) {
			was_error = 1;
			goto done;
			}
			IMA_ADPCM_encoded = 1;
			break;
			case EXTENSIBLE_CODE:
			/* note that this ignores channel masks, smaller valid bit counts
			inside a larger container, and most subtypes. This is just enough
			to get things that didn't really _need_ WAVE_FORMAT_EXTENSIBLE
			to be useful working when they use this format flag. */
			ext = (WaveExtensibleFMT *) format;
			if (SDL_SwapLE16(ext->size) < 22) {
			SDL_SetError("bogus extended .wav header");
			was_error = 1;
			goto done;
			}
			if (SDL_memcmp(ext->subformat, extensible_pcm_guid, 16) == 0) {
			break; /* cool. */
			} else if (SDL_memcmp(ext->subformat, extensible_ieee_guid, 16) == 0) {
			IEEE_float_encoded = 1;
			break;
			if (IMA_ADPCM_Init(file, datalength) < 0) {
			return -1;
			}
			break;
			case MP3_CODE:
			SDL_SetError("MPEG Layer 3 data not supported");
			was_error = 1;
			goto done;
			case MPEG_CODE:
			case MPEGLAYER3_CODE:
			return SDL_SetError("MPEG formats not supported");
			default:
			SDL_SetError("Unknown WAVE data format: 0x%.4x",
			SDL_SwapLE16(format->encoding));
			was_error = 1;
			goto done;
			if (format->formattag == EXTENSIBLE_CODE) {
			const char *errstr = "Unknown WAVE format GUID: %08x-%04x-%04x-%02x%02x%02x%02x%02x%02x%02x%02x";
			const Uint8 *g = format->subformat;
			const Uint32 g1 = g[0] \| ((Uint32)g[1] << 8) \| ((Uint32)g[2] << 16) \| ((Uint32)g[3] << 24);
			const Uint32 g2 = g[4] \| ((Uint32)g[5] << 8);
			const Uint32 g3 = g[6] \| ((Uint32)g[7] << 8);
			return SDL_SetError(errstr, g1, g2, g3, g[8], g[9], g[10], g[11], g[12], g[13], g[14], g[15]);
			}
			SDL_zerop(spec);
			spec->freq = SDL_SwapLE32(format->frequency);
			return SDL_SetError("Unknown WAVE format tag: 0x%04x", (unsigned int)format->encoding);
			}

			if (IEEE_float_encoded) {
			if ((SDL_SwapLE16(format->bitspersample)) != 32) {
			was_error = 1;
			} else {
			spec->format = AUDIO_F32;
			return 0;
			}

			static int
			WaveLoad(SDL_RWops src, WaveFile file, SDL_AudioSpec spec, Uint8 audio_buf, Uint32 audio_len)
			{
			int result;
			Uint32 chunkcount = 0;
			Uint32 chunkcountlimit = 10000;
			char *envchunkcountlimit;
			Sint64 RIFFstart, RIFFend, lastchunkpos;
			SDL_bool RIFFlengthknown = SDL_FALSE;
			WaveFormat *format = &file->format;
			WaveChunk *chunk = &file->chunk;
			WaveChunk RIFFchunk;
			WaveChunk fmtchunk;
			WaveChunk datachunk;

			SDL_zero(RIFFchunk);
			SDL_zero(fmtchunk);
			SDL_zero(datachunk);

			envchunkcountlimit = SDL_getenv("SDL_WAVE_CHUNK_LIMIT");
			if (envchunkcountlimit != NULL) {
			unsigned int count;
			if (SDL_sscanf(envchunkcountlimit, "%u", &count) == 1) {
			chunkcountlimit = count <= SDL_MAX_UINT32 ? count : SDL_MAX_UINT32;
			}
			}

			RIFFstart = SDL_RWtell(src);
			if (RIFFstart < 0) {
			return SDL_SetError("Could not seek in file");
			}

			RIFFchunk.position = RIFFstart;
			if (WaveNextChunk(src, &RIFFchunk) < 0) {
			return SDL_SetError("Could not read RIFF header");
			}

			/* Check main WAVE file identifiers. */
			if (RIFFchunk.fourcc == RIFF) {
			Uint32 formtype;
			/* Read the form type. "WAVE" expected. */
			if (SDL_RWread(src, &formtype, sizeof(Uint32), 1) != 1) {
			return SDL_SetError("Could not read RIFF form type");
			} else if (SDL_SwapLE32(formtype) != WAVE) {
			return SDL_SetError("RIFF form type is not WAVE (not a Waveform file)");
			}
			} else if (RIFFchunk.fourcc == WAVE) {
			/* RIFF chunk missing or skipped. Length unknown. */
			RIFFchunk.position = 0;
			RIFFchunk.length = 0;
			} else {
			switch (SDL_SwapLE16(format->bitspersample)) {
			case 4:
			if (MS_ADPCM_encoded \|\| IMA_ADPCM_encoded) {
			spec->format = AUDIO_S16;
			return SDL_SetError("Could not find RIFF or WAVE identifiers (not a Waveform file)");
			}

			/* The 4-byte form type is immediately followed by the first chunk.*/
			chunk->position = RIFFchunk.position + 4;

			/* Use the RIFF chunk size to limit the search for the chunks. This is not
			* always reliable and the hint can be used to tune the behavior. By
			* default, it will never search past 4 GiB.
			*/
			switch (file->riffhint) {
			case RiffSizeIgnore:
			RIFFend = RIFFchunk.position + SDL_MAX_UINT32;
			break;
			default:
			case RiffSizeIgnoreZero:
			if (RIFFchunk.length == 0) {
			RIFFend = RIFFchunk.position + SDL_MAX_UINT32;
			break;
			}
			/* fallthrough */
			case RiffSizeForce:
			RIFFend = RIFFchunk.position + RIFFchunk.length;
			RIFFlengthknown = SDL_TRUE;
			break;
			case RiffSizeMaximum:
			RIFFend = SDL_MAX_SINT64;
			break;
			}

			/* Step through all chunks and save information on the fmt, data, and fact
			* chunks. Ignore the chunks we don't know as per specification. This
			* currently also ignores cue, list, and slnt chunks.
			*/
			while ((Uint64)RIFFend > (Uint64)chunk->position + chunk->length + (chunk->length & 1)) {
			/* Abort after too many chunks or else corrupt files may waste time. */
			if (chunkcount++ >= chunkcountlimit) {
			return SDL_SetError("Chunk count in WAVE file exceeds limit of %u", chunkcountlimit);
			}

			result = WaveNextChunk(src, chunk);
			if (result == -1) {
			/* Unexpected EOF. Corrupt file or I/O issues. */
			if (file->trunchint == TruncVeryStrict) {
			return SDL_SetError("Unexpected end of WAVE file");
			}
			/* Let the checks after this loop sort this issue out. */
			break;
			} else if (result == -2) {
			return SDL_SetError("Could not seek to WAVE chunk header");
			}

			if (chunk->fourcc == FMT) {
			if (fmtchunk.fourcc == FMT) {
			/* Multiple fmt chunks. Ignore or error? */
			} else {
			was_error = 1;
			/* The fmt chunk must occur before the data chunk. */
			if (datachunk.fourcc == DATA) {
			return SDL_SetError("fmt chunk after data chunk in WAVE file");
			}
			fmtchunk = *chunk;
			}
			} else if (chunk->fourcc == DATA) {
			/* Only use the first data chunk. Handling the wavl list madness
			* may require a different approach.
			*/
			if (datachunk.fourcc != DATA) {
			datachunk = *chunk;
			}
			} else if (chunk->fourcc == FACT) {
			/* The fact chunk data must be at least 4 bytes for the
			* dwSampleLength field. Ignore all fact chunks after the first one.
			*/
			if (file->fact.status == 0) {
			if (chunk->length < 4) {
			file->fact.status = -1;
			} else {
			/* Let's use src directly, it's just too convenient. */
			Sint64 position = SDL_RWseek(src, chunk->position, RW_SEEK_SET);
			Uint32 samplelength;
			if (position == chunk->position && SDL_RWread(src, &samplelength, sizeof(Uint32), 1) == 1) {
			file->fact.status = 1;
			file->fact.samplelength = SDL_SwapLE32(samplelength);
			} else {
			file->fact.status = -1;
			}
			}
			}
			}

			/* Go through all chunks in verystrict mode or stop the search early if
			* all required chunks were found.
			*/
			if (file->trunchint == TruncVeryStrict) {
			if ((Uint64)RIFFend < (Uint64)chunk->position + chunk->length) {
			return SDL_SetError("RIFF size truncates chunk");
			}
			} else if (fmtchunk.fourcc == FMT && datachunk.fourcc == DATA) {
			if (file->fact.status == 1 \|\| file->facthint == FactIgnore \|\| file->facthint == FactNoHint) {
			break;
			}
			}
			}

			/* Save the position after the last chunk. This position will be used if the
			* RIFF length is unknown.
			*/
			lastchunkpos = chunk->position + chunk->length;

			/* The fmt chunk is mandatory. */
			if (fmtchunk.fourcc != FMT) {
			return SDL_SetError("Missing fmt chunk in WAVE file");
			}
			/* A data chunk must be present. */
			if (datachunk.fourcc != DATA) {
			return SDL_SetError("Missing data chunk in WAVE file");
			}
			/* Check if the last chunk has all of its data in verystrict mode. */
			if (file->trunchint == TruncVeryStrict) {
			/* data chunk is handled later. */
			if (chunk->fourcc != DATA && chunk->length > 0) {
			Uint8 tmp;
			Uint64 position = (Uint64)chunk->position + chunk->length - 1;
			if (position > SDL_MAX_SINT64 \|\| SDL_RWseek(src, (Sint64)position, RW_SEEK_SET) != (Sint64)position) {
			return SDL_SetError("Could not seek to WAVE chunk data");
			} else if (SDL_RWread(src, &tmp, 1, 1) != 1) {
			return SDL_SetError("RIFF size truncates chunk");
			}
			}
			}

			/* Process fmt chunk. */
			*chunk = fmtchunk;

			/* No need to read more than 1046 bytes of the fmt chunk data with the
			* formats that are currently supported. (1046 because of MS ADPCM coefficients)
			*/
			if (WaveReadPartialChunkData(src, chunk, 1046) < 0) {
			return SDL_SetError("Could not read data of WAVE fmt chunk");
			}

			/* The fmt chunk data must be at least 14 bytes to include all common fields.
			* It usually is 16 and larger depending on the header and encoding.
			*/
			if (chunk->length < 14) {
			return SDL_SetError("Invalid WAVE fmt chunk length (too small)");
			} else if (chunk->size < 14) {
			return SDL_SetError("Could not read data of WAVE fmt chunk");
			} else if (WaveReadFormat(file) < 0) {
			return -1;
			} else if (WaveCheckFormat(file, (size_t)datachunk.length) < 0) {
			return -1;
			}

			#ifdef SDL_WAVE_DEBUG_LOG_FORMAT
			WaveDebugLogFormat(file);
			#endif
			#ifdef SDL_WAVE_DEBUG_DUMP_FORMAT
			WaveDebugDumpFormat(file, RIFFchunk.length, fmtchunk.length, datachunk.length);
			#endif

			WaveFreeChunkData(chunk);

			/* Process data chunk. */
			*chunk = datachunk;

			if (chunk->length > 0) {
			result = WaveReadChunkData(src, chunk);
			if (result == -1) {
			return -1;
			} else if (result == -2) {
			return SDL_SetError("Could not seek data of WAVE data chunk");
			}
			}

			if (chunk->length != chunk->size) {
			/* I/O issues or corrupt file. */
			if (file->trunchint == TruncVeryStrict \|\| file->trunchint == TruncStrict) {
			return SDL_SetError("Could not read data of WAVE data chunk");
			}
			/* The decoders handle this truncation. */
			}

			/* Decode or convert the data if necessary. */
			switch (format->encoding) {
			case PCM_CODE:
			case IEEE_FLOAT_CODE:
			if (PCM_Decode(file, audio_buf, audio_len) < 0) {
			return -1;
			}
			break;
			case ALAW_CODE:
			case MULAW_CODE:
			if (LAW_Decode(file, audio_buf, audio_len) < 0) {
			return -1;
			}
			break;
			case MS_ADPCM_CODE:
			if (MS_ADPCM_Decode(file, audio_buf, audio_len) < 0) {
			return -1;
			}
			break;
			case IMA_ADPCM_CODE:
			if (IMA_ADPCM_Decode(file, audio_buf, audio_len) < 0) {
			return -1;
			}
			break;
			}

			/* Setting up the SDL_AudioSpec. 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 */

			switch (format->encoding) {
			case MS_ADPCM_CODE:
			case IMA_ADPCM_CODE:
			case ALAW_CODE:
			case MULAW_CODE:
			/* These can be easily stored in the byte order of the system. */
			spec->format = AUDIO_S16SYS;
			break;
			case IEEE_FLOAT_CODE:
			spec->format = AUDIO_F32LSB;
			break;
			case PCM_CODE:
			switch (format->bitspersample) {
			case 8:
			spec->format = AUDIO_U8;
			break;
			case 16:
			spec->format = AUDIO_S16;
			spec->format = AUDIO_S16LSB;
			break;
			case 24: /* convert this. */
			spec->format = AUDIO_S32;
			break;
			case 24: /* Has been shifted to 32 bits. */
			case 32:
			spec->format = AUDIO_S32;
			spec->format = AUDIO_S32LSB;
			break;
			default:
			was_error = 1;
			/* Just in case something unexpected happened in the checks. */
			return SDL_SetError("Unexpected %u-bit PCM data format", (unsigned int)format->bitspersample);
			}
			break;
			}

			spec->silence = SDL_SilenceValueForFormat(spec->format);

			/* Report the end position back to the cleanup code. */
			if (RIFFlengthknown) {
			chunk->position = RIFFend;
			} else {
			chunk->position = lastchunkpos;
			}

			if (was_error) {
			SDL_SetError("Unknown %d-bit PCM data format",
			SDL_SwapLE16(format->bitspersample));
			goto done;
			return 0;
			}
			spec->channels = (Uint8) SDL_SwapLE16(format->channels);
			spec->samples = 4096; /* Good default buffer size */

			/* Read the audio data chunk */
			SDL_AudioSpec *
			SDL_LoadWAV_RW(SDL_RWops src, int freesrc, SDL_AudioSpec spec, Uint8 *audio_buf, Uint32 audio_len)
			{
			int result;
			WaveFile file;

			SDL_zero(file);

			/* Make sure we are passed a valid data source */
			if (src == NULL) {
			/* Error may come from RWops. */
			return NULL;
			} else if (spec == NULL) {
			SDL_InvalidParamError("spec");
			return NULL;
			} else if (audio_buf == NULL) {
			SDL_InvalidParamError("audio_buf");
			return NULL;
			} else if (audio_len == NULL) {
			SDL_InvalidParamError("audio_len");
			return NULL;
			}

			*audio_buf = NULL;
			do {
			*audio_len = 0;

			file.riffhint = WaveGetRiffSizeHint();
			file.trunchint = WaveGetTruncationHint();
			file.facthint = WaveGetFactChunkHint();

			result = WaveLoad(src, &file, spec, audio_buf, audio_len);
			if (result < 0) {
			SDL_free(*audio_buf);
			*audio_buf = NULL;
			lenread = ReadChunk(src, &chunk);
			if (lenread < 0) {
			was_error = 1;
			goto done;
			}
			*audio_len = lenread;
			*audio_buf = chunk.data;
			if (chunk.magic != DATA)
			headerDiff += lenread + 2 * sizeof(Uint32);
			} while (chunk.magic != DATA);
			headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */

			if (MS_ADPCM_encoded) {
			if (MS_ADPCM_decode(audio_buf, audio_len) < 0) {
			was_error = 1;
			goto done;
			}
			}
			if (IMA_ADPCM_encoded) {
			if (IMA_ADPCM_decode(audio_buf, audio_len) < 0) {
			was_error = 1;
			goto done;
			}
			spec = NULL;
			audio_buf = NULL;
			audio_len = 0;
			}

			if (SDL_SwapLE16(format->bitspersample) == 24) {
			if (ConvertSint24ToSint32(audio_buf, audio_len) < 0) {
			was_error = 1;
			goto done;
			}
			}

			/* Don't return a buffer that isn't a multiple of samplesize */
			samplesize = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
			*audio_len &= ~(samplesize - 1);

			done:
			SDL_free(format);
			if (src) {
			/* Cleanup */
			if (freesrc) {
			SDL_RWclose(src);
			} else {
			/* seek to the end of the file (given by the RIFF chunk) */
			SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
			SDL_RWseek(src, file.chunk.position, RW_SEEK_SET);
			}
			}
			if (was_error) {
			spec = NULL;
			}
			return (spec);
			WaveFreeChunkData(&file.chunk);
			SDL_free(file.decoderdata);

			return spec;
			}

			/* Since the WAV memory is allocated in the shared library, it must also
			@@ -672,23 +2151,6 @@
			SDL_FreeWAV(Uint8 * audio_buf)
			{
			SDL_free(audio_buf);
			}

			static int
			ReadChunk(SDL_RWops * src, Chunk * chunk)
			{
			chunk->magic = SDL_ReadLE32(src);
			chunk->length = SDL_ReadLE32(src);
			chunk->data = (Uint8 *) SDL_malloc(chunk->length);
			if (chunk->data == NULL) {
			return SDL_OutOfMemory();
			}
			if (SDL_RWread(src, chunk->data, chunk->length, 1) != 1) {
			SDL_free(chunk->data);
			chunk->data = NULL;
			return SDL_Error(SDL_EFREAD);
			}
			return (chunk->length);
			}

			/* vi: set ts=4 sw=4 expandtab: */