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SUSE:SLE-15-SP3:GA
SDL2.22370
CVE-2019-13626.patch
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File CVE-2019-13626.patch of Package SDL2.22370
diff -urp SDL2-2.0.8.orig/include/SDL_audio.h SDL2-2.0.8/include/SDL_audio.h --- SDL2-2.0.8.orig/include/SDL_audio.h 2018-03-01 10:34:41.000000000 -0600 +++ SDL2-2.0.8/include/SDL_audio.h 2019-08-26 10:45:44.192574222 -0500 @@ -419,23 +419,56 @@ extern DECLSPEC void SDLCALL SDL_PauseAu /* @} *//* Pause audio functions */ /** - * This function loads a WAVE from the data source, automatically freeing - * that source if \c freesrc is non-zero. For example, to load a WAVE file, - * you could do: + * \brief Load the audio data of a WAVE file into memory + * + * Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len + * to be valid pointers. The entire data portion of the file is then loaded + * into memory and decoded if necessary. + * + * If \c freesrc is non-zero, the data source gets automatically closed and + * freed before the function returns. + * + * Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits), + * IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and + * ยต-law (8 bits). Other formats are currently unsupported and cause an error. + * + * If this function succeeds, the pointer returned by it is equal to \c spec + * and the pointer to the audio data allocated by the function is written to + * \c audio_buf and its length in bytes to \c audio_len. The \ref SDL_AudioSpec + * members \c freq, \c channels, and \c format are set to the values of the + * audio data in the buffer. The \c samples member is set to a sane default and + * all others are set to zero. + * + * It's necessary to use SDL_FreeWAV() to free the audio data returned in + * \c audio_buf when it is no longer used. + * + * Because of the underspecification of the Waveform format, there are many + * problematic files in the wild that cause issues with strict decoders. To + * provide compatibility with these files, this decoder is lenient in regards + * to the truncation of the file, the fact chunk, and the size of the RIFF + * chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE, SDL_HINT_WAVE_TRUNCATION, + * and SDL_HINT_WAVE_FACT_CHUNK can be used to tune the behavior of the + * loading process. + * + * Any file that is invalid (due to truncation, corruption, or wrong values in + * the headers), too big, or unsupported causes an error. Additionally, any + * critical I/O error from the data source will terminate the loading process + * with an error. The function returns NULL on error and in all cases (with the + * exception of \c src being NULL), an appropriate error message will be set. + * + * It is required that the data source supports seeking. + * + * Example: * \code * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); * \endcode * - * If this function succeeds, it returns the given SDL_AudioSpec, - * filled with the audio data format of the wave data, and sets - * \c *audio_buf to a malloc()'d buffer containing the audio data, - * and sets \c *audio_len to the length of that audio buffer, in bytes. - * You need to free the audio buffer with SDL_FreeWAV() when you are - * done with it. - * - * This function returns NULL and sets the SDL error message if the - * wave file cannot be opened, uses an unknown data format, or is - * corrupt. Currently raw and MS-ADPCM WAVE files are supported. + * \param src The data source with the WAVE data + * \param freesrc A integer value that makes the function close the data source if non-zero + * \param spec A pointer filled with the audio format of the audio data + * \param audio_buf A pointer filled with the audio data allocated by the function + * \param audio_len A pointer filled with the length of the audio data buffer in bytes + * \return NULL on error, or non-NULL on success. */ extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, int freesrc, Only in SDL2-2.0.8/include: SDL_audio.h.orig diff -urp SDL2-2.0.8.orig/include/SDL_hints.h SDL2-2.0.8/include/SDL_hints.h --- SDL2-2.0.8.orig/include/SDL_hints.h 2018-03-01 10:34:41.000000000 -0600 +++ SDL2-2.0.8/include/SDL_hints.h 2019-08-26 10:45:44.192574222 -0500 @@ -931,6 +931,70 @@ extern "C" { #define SDL_HINT_AUDIO_CATEGORY "SDL_AUDIO_CATEGORY" /** + * \brief Controls how the size of the RIFF chunk affects the loading of a WAVE file. + * + * The size of the RIFF chunk (which includes all the sub-chunks of the WAVE + * file) is not always reliable. In case the size is wrong, it's possible to + * just ignore it and step through the chunks until a fixed limit is reached. + * + * Note that files that have trailing data unrelated to the WAVE file or + * corrupt files may slow down the loading process without a reliable boundary. + * By default, SDL stops after 10000 chunks to prevent wasting time. Use the + * environment variable SDL_WAVE_CHUNK_LIMIT to adjust this value. + * + * This variable can be set to the following values: + * + * "chunksearch" - Use the RIFF chunk size as a boundary for the chunk search + * "ignorezero" - Like "chunksearch", but a zero size searches up to 4 GiB (default) + * "ignore" - Ignore the RIFF chunk size and always search up to 4 GiB + * "maximum" - Search for chunks until the end of file (not recommended) + */ +#define SDL_HINT_WAVE_RIFF_CHUNK_SIZE "SDL_WAVE_RIFF_CHUNK_SIZE" + +/** + * \brief Controls how a truncated WAVE file is handled. + * + * A WAVE file is considered truncated if any of the chunks are incomplete or + * the data chunk size is not a multiple of the block size. By default, SDL + * decodes until the first incomplete block, as most applications seem to do. + * + * This variable can be set to the following values: + * + * "verystrict" - Raise an error if the file is truncated + * "strict" - Like "verystrict", but the size of the RIFF chunk is ignored + * "dropframe" - Decode until the first incomplete sample frame + * "dropblock" - Decode until the first incomplete block (default) + */ +#define SDL_HINT_WAVE_TRUNCATION "SDL_WAVE_TRUNCATION" + +/** + * \brief Controls how the fact chunk affects the loading of a WAVE file. + * + * The fact chunk stores information about the number of samples of a WAVE + * file. The Standards Update from Microsoft notes that this value can be used + * to 'determine the length of the data in seconds'. This is especially useful + * for compressed formats (for which this is a mandatory chunk) if they produce + * multiple sample frames per block and truncating the block is not allowed. + * The fact chunk can exactly specify how many sample frames there should be + * in this case. + * + * Unfortunately, most application seem to ignore the fact chunk and so SDL + * ignores it by default as well. + * + * This variable can be set to the following values: + * + * "truncate" - Use the number of samples to truncate the wave data if + * the fact chunk is present and valid + * "strict" - Like "truncate", but raise an error if the fact chunk + * is invalid, not present for non-PCM formats, or if the + * data chunk doesn't have that many samples + * "ignorezero" - Like "truncate", but ignore fact chunk if the number of + * samples is zero + * "ignore" - Ignore fact chunk entirely (default) + */ +#define SDL_HINT_WAVE_FACT_CHUNK "SDL_WAVE_FACT_CHUNK" + +/** * \brief An enumeration of hint priorities */ typedef enum Only in SDL2-2.0.8/include: SDL_hints.h.orig diff -urp SDL2-2.0.8.orig/src/audio/SDL_wave.c SDL2-2.0.8/src/audio/SDL_wave.c --- SDL2-2.0.8.orig/src/audio/SDL_wave.c 2018-03-01 10:34:42.000000000 -0600 +++ SDL2-2.0.8/src/audio/SDL_wave.c 2019-08-26 10:45:44.192574222 -0500 @@ -1,6 +1,6 @@ /* Simple DirectMedia Layer - Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org> + Copyright (C) 1997-2019 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 +20,849 @@ */ #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" +/* Reads the value stored at the location of the f1 pointer, multiplies it + * with the second argument, and then stores it back to f1 again. + * Returns SDL_TRUE if the multiplication overflows, f1 does not get modified. + */ +static SDL_bool +MultiplySize(size_t *f1, size_t f2) +{ + if (*f1 > 0 && SIZE_MAX / *f1 <= f2) { + return SDL_TRUE; + } + *f1 *= f2; + return SDL_FALSE; +} -static int ReadChunk(SDL_RWops * src, Chunk * chunk); +typedef struct ADPCM_DecoderState +{ + 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; -struct MS_ADPCM_decodestate +typedef struct MS_ADPCM_CoeffData { - Uint8 hPredictor; - Uint16 iDelta; - Sint16 iSamp1; - Sint16 iSamp2; -}; -static struct MS_ADPCM_decoder + 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) { - WaveFMT wavefmt; - Uint16 wSamplesPerBlock; - Uint16 wNumCoef; - Sint16 aCoeff[7][2]; - /* * * */ - struct MS_ADPCM_decodestate state[2]; -} MS_ADPCM_state; - -static int -InitMS_ADPCM(WaveFMT * format) -{ - Uint8 *rogue_feel; - int i; - - /* 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); - } - 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) -{ - const Sint32 max_audioval = ((1 << (16 - 1)) - 1); - const Sint32 min_audioval = -(1 << (16 - 1)); - const Sint32 adaptive[] = { + 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] = {0}; + + 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 +MS_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength) +{ + WaveFormat *format = &file->format; + const size_t blockheadersize = file->format.channels * 7; + const size_t availableblocks = datalength / file->format.blockalign; + const size_t blockframebitsize = file->format.bitspersample * file->format.channels; + const size_t trailingdata = datalength % file->format.blockalign; + + 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"); + } + } + + /* 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) +{ + WaveFormat *format = &file->format; + WaveChunk *chunk = &file->chunk; + const size_t blockheadersize = format->channels * 7; + const size_t blockdatasize = (size_t)format->blockalign - blockheadersize; + const size_t blockframebitsize = 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 %d", (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; + + 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); - /* 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) { - 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]; - - /* 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 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; + /* 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; } - encoded_len -= MS_ADPCM_state.wavefmt.blockalign; + 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++; } - SDL_free(freeable); - return (0); + + 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; } -struct IMA_ADPCM_decodestate +/* 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) { - Sint32 sample; - Sint8 index; -}; -static struct IMA_ADPCM_decoder + 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 & 0x8000) { + nybble <<= 4; + } else if (blockpos < blocksize) { + nybble = state->block.data[blockpos++] | 0x8000; + } 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 = {0}; + MS_ADPCM_ChannelState cstate[2] = {0}; + + 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 = 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 (MultiplySize(&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(); + } + + state.cstate = &cstate; + + /* 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"); + } + + /* Initialize decoder with the values from the block header. */ + result = MS_ADPCM_DecodeBlockHeader(&state); + if (result == -1) { + SDL_free(state.output.data); + return -1; + } + + /* 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 == TruncVeryStrict) { + 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 +IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength) { - WaveFMT wavefmt; - Uint16 wSamplesPerBlock; - /* * * */ - struct IMA_ADPCM_decodestate state[2]; -} IMA_ADPCM_state; - -static int -InitIMA_ADPCM(WaveFMT * format) -{ - Uint8 *rogue_feel; - - /* 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); - } - 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) -{ - const Sint32 max_audioval = ((1 << (16 - 1)) - 1); - const Sint32 min_audioval = -(1 << (16 - 1)); - const int index_table[16] = { + WaveFormat *format = &file->format; + const size_t blockheadersize = format->channels * 4; + const size_t subblockframesize = format->channels * 4; + const size_t availableblocks = datalength / format->blockalign; + const size_t trailingdata = datalength % format->blockalign; + + 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"); + } + } + + /* 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) +{ + WaveFormat *format = &file->format; + WaveChunk *chunk = &file->chunk; + const size_t blockheadersize = format->channels * 4; + const size_t blockdatasize = (size_t)format->blockalign - blockheadersize; + const size_t blockframebitsize = format->bitspersample * format->channels; + const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize; + + /* Sanity checks. */ + + /* IMA ADPCAM 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 %d", (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, @@ -271,424 +872,1260 @@ IMA_ADPCM_nibble(struct IMA_ADPCM_decode 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 }; - Sint32 delta, step; - - /* Compute difference and new sample value */ - if (state->index > 88) { - state->index = 88; - } else if (state->index < 0) { - state->index = 0; + Uint32 step; + Sint32 sample, delta; + Sint8 index = *cindex; + + /* 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; + if (sample > max_audioval) { + sample = max_audioval; + } else if (sample < min_audioval) { + sample = min_audioval; } - return (state->sample); + + return (Sint16)sample; } -/* 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) +static int +IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state) { - int i; - Sint8 nybble; - Sint32 new_sample; + Sint16 step; + Uint32 c; + Uint8 *cstate = state->cstate; - 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; + for (c = 0; c < state->channels; c++) { + size_t o = state->block.pos + c * 4; - 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; + /* Extract the sample from the header. */ + Sint32 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; - ++encoded; + /* 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? */ ; + } } + + 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_decode(Uint8 ** audio_buf, Uint32 * audio_len) +IMA_ADPCM_DecodeBlockData(ADPCM_DecoderState *state) { - struct IMA_ADPCM_decodestate *state; - Uint8 *freeable, *encoded, *decoded; - Sint32 encoded_len, samplesleft; - unsigned int c, channels; + 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; + } + } - /* 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); + outpos += channels * subblocksamples; + state->framesleft -= subblocksamples; + blockframesleft -= subblocksamples; } - state = IMA_ADPCM_state.state; - /* 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) { + state->block.pos = blockpos; + state->output.pos = outpos; + + return retval; +} + +static int +IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len) +{ + int result; + size_t bytesleft, outputsize; + WaveChunk *chunk = &file->chunk; + ADPCM_DecoderState state = {0}; + Sint8 *cstate; + + 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; + } + + state.channels = file->format.channels; + state.blocksize = file->format.blockalign; + state.blockheadersize = 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 (MultiplySize(&outputsize, state.framesize)) { return SDL_OutOfMemory(); + } else if (outputsize > SDL_MAX_UINT32 || state.framestotal > SIZE_MAX) { + return SDL_SetError("WAVE file too big"); } - decoded = *audio_buf; - /* 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) { - /* Uh oh, corrupt data? Buggy code? */ ; + 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(); + } + + cstate = (Sint8 *)SDL_calloc(state.channels, sizeof(Sint8)); + if (cstate == NULL) { + SDL_free(state.output.data); + return SDL_OutOfMemory(); + } + state.cstate = cstate; + + /* 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); + + /* 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 == TruncVeryStrict) { + 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 %d", (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"); + } + } - /* Store the initial sample we start with */ - decoded[0] = (Uint8) (state[c].sample & 0xFF); - decoded[1] = (Uint8) (state[c].sample >> 8); - decoded += 2; + file->sampleframes = WaveAdjustToFactValue(file, datalength / format->blockalign); + if (file->sampleframes < 0) { + return -1; + } + + return 0; +} + +static int +LAW_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len) +{ +#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 + + WaveFormat *format = &file->format; + WaveChunk *chunk = &file->chunk; + size_t i, sample_count, expanded_len; + Uint8 *src; + Sint16 *dst; + + 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; + } + + sample_count = (size_t)file->sampleframes; + if (MultiplySize(&sample_count, format->channels)) { + return SDL_OutOfMemory(); + } + + expanded_len = sample_count; + if (MultiplySize(&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"); + } + + src = (Uint8 *)SDL_realloc(chunk->data, expanded_len); + if (src == NULL) { + return SDL_OutOfMemory(); + } + chunk->data = NULL; + chunk->size = 0; + + dst = (Sint16 *)src; - /* 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; + /* 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]]; + } + 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; } - decoded += (channels * 8 * 2); + mantissa = mantissa << 4 | 0x8; + if (exponent > 1) { + mantissa <<= exponent - 1; + } + + dst[i] = nibble & 0x80 ? mantissa : -mantissa; } - encoded_len -= IMA_ADPCM_state.wavefmt.blockalign; + 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"); } - SDL_free(freeable); - return (0); + + *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("%d-bit PCM format not supported", (int)format->bitspersample); + } + } else if (format->encoding == IEEE_FLOAT_CODE) { + if (format->bitspersample != 32) { + return SDL_SetError("%d-bit IEEE floating-point format not supported", (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 -ConvertSint24ToSint32(Uint8 ** audio_buf, Uint32 * audio_len) +PCM_ConvertSint24ToSint32(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; + WaveFormat *format = &file->format; + WaveChunk *chunk = &file->chunk; + size_t i, expanded_len, sample_count; + Uint8 *ptr; + + sample_count = (size_t)file->sampleframes; + if (MultiplySize(&sample_count, format->channels)) { + return SDL_OutOfMemory(); + } - if (!ptr) { + expanded_len = sample_count; + if (MultiplySize(&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"); } + ptr = (Uint8 *)SDL_realloc(chunk->data, expanded_len); + if (ptr == NULL) { + return SDL_OutOfMemory(); + } + + /* This pointer is now invalid. */ + chunk->data = NULL; + chunk->size = 0; + *audio_buf = ptr; - *audio_len = expanded_len; + *audio_len = (Uint32)expanded_len; /* 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); + 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; -/* 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 }; + if (chunk->length != chunk->size) { + file->sampleframes = WaveAdjustToFactValue(file, chunk->size / format->blockalign); + if (file->sampleframes < 0) { + return -1; + } + } -SDL_AudioSpec * -SDL_LoadWAV_RW(SDL_RWops * src, int freesrc, - SDL_AudioSpec * spec, Uint8 ** audio_buf, Uint32 * audio_len) + /* 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 (MultiplySize(&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() { - int was_error; - Chunk chunk; - int lenread; - int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded; - int samplesize; - - /* WAV magic header */ - Uint32 RIFFchunk; - Uint32 wavelen = 0; - Uint32 WAVEmagic; - Uint32 headerDiff = 0; - - /* FMT chunk */ - WaveFMT *format = NULL; - WaveExtensibleFMT *ext = NULL; + const char *hint = SDL_GetHint(SDL_HINT_WAVE_RIFF_CHUNK_SIZE); - SDL_zero(chunk); + if (hint != NULL) { + if (SDL_strcmp(hint, "chunksearch") == 0) { + return RiffSizeChunkSearch; + } 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; + } + } - /* Make sure we are passed a valid data source */ - was_error = 0; - if (src == NULL) { - was_error = 1; - goto done; + 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; + } } - /* 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); + 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; + } } - if ((RIFFchunk != RIFF) || (WAVEmagic != WAVE)) { - SDL_SetError("Unrecognized file type (not WAVE)"); - was_error = 1; - goto done; - } - headerDiff += sizeof(Uint32); /* for WAVE */ - - /* 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; - } - /* 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; + + 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); + + /* 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(); } - IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0; - switch (SDL_SwapLE16(format->encoding)) { + + 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 the 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: + 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: - /* We can understand this */ - break; case IEEE_FLOAT_CODE: - IEEE_float_encoded = 1; - /* We can understand this */ + 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; + if (IMA_ADPCM_Init(file, datalength) < 0) { + return -1; } - 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; + case MPEG_CODE: + case MPEGLAYER3_CODE: + return SDL_SetError("MPEG formats not supported"); + default: + 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]); } + return SDL_SetError("Unknown WAVE format tag: 0x%04x", (int)format->encoding); + } + + 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 = {0}; + WaveChunk fmtchunk = {0}; + WaveChunk datachunk = {0}; + + 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 { + 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; - case MP3_CODE: - SDL_SetError("MPEG Layer 3 data not supported"); - was_error = 1; - goto done; default: - SDL_SetError("Unknown WAVE data format: 0x%.4x", - SDL_SwapLE16(format->encoding)); - was_error = 1; - goto done; + case RiffSizeIgnoreZero: + if (RIFFchunk.length == 0) { + RIFFend = RIFFchunk.position + SDL_MAX_UINT32; + break; + } + /* fallthrough */ + case RiffSizeChunkSearch: + RIFFend = RIFFchunk.position + RIFFchunk.length; + RIFFlengthknown = SDL_TRUE; + break; + case RiffSizeMaximum: + RIFFend = SDL_MAX_SINT64; + break; } - SDL_zerop(spec); - spec->freq = SDL_SwapLE32(format->frequency); - if (IEEE_float_encoded) { - if ((SDL_SwapLE16(format->bitspersample)) != 32) { - was_error = 1; - } else { - spec->format = AUDIO_F32; + /* 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 (RIFFend > 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); } - } else { - switch (SDL_SwapLE16(format->bitspersample)) { - case 4: - if (MS_ADPCM_encoded || IMA_ADPCM_encoded) { - spec->format = AUDIO_S16; - } else { - was_error = 1; + + 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 { + /* 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 (RIFFend < 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; + Sint64 position = chunk->position + chunk->length - 1; + if (SDL_RWseek(src, position, RW_SEEK_SET) != 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; - break; - case 24: /* convert this. */ - spec->format = AUDIO_S32; + spec->format = AUDIO_S16LSB; break; + case 24: /* Has been shifted to 32 bits. */ case 32: - spec->format = AUDIO_S32; + spec->format = AUDIO_S32LSB; break; default: - was_error = 1; - break; + /* Just in case something unexpected happened in the checks. */ + return SDL_SetError("Unexpected %d-bit PCM data format", format->bitspersample); } + break; } - if (was_error) { - SDL_SetError("Unknown %d-bit PCM data format", - SDL_SwapLE16(format->bitspersample)); - goto done; + /* Report the end position back to the cleanup code. */ + if (RIFFlengthknown) { + chunk->position = RIFFend; + } else { + chunk->position = lastchunkpos; } - spec->channels = (Uint8) SDL_SwapLE16(format->channels); - spec->samples = 4096; /* Good default buffer size */ - /* Read the audio data chunk */ - *audio_buf = NULL; - do { - 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 */ + return 0; +} - 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; - } - } +SDL_AudioSpec * +SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len) +{ + int result; + WaveFile file = {0}; - if (SDL_SwapLE16(format->bitspersample) == 24) { - if (ConvertSint24ToSint32(audio_buf, audio_len) < 0) { - was_error = 1; - goto done; - } + /* 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; } - /* 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); + *audio_buf = NULL; + *audio_len = 0; + + file.riffhint = WaveGetRiffSizeHint(); + file.trunchint = WaveGetTruncationHint(); + file.facthint = WaveGetFactChunkHint(); - done: - SDL_free(format); - if (src) { - 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); - } - } - if (was_error) { + result = WaveLoad(src, &file, spec, audio_buf, audio_len); + if (result < 0) { + SDL_free(*audio_buf); spec = NULL; + audio_buf = NULL; + audio_len = 0; } - return (spec); + + /* Cleanup */ + if (freesrc) { + SDL_RWclose(src); + } else { + SDL_RWseek(src, file.chunk.position, RW_SEEK_SET); + } + WaveFreeChunkData(&file.chunk); + SDL_free(file.decoderdata); + + return spec; } /* Since the WAV memory is allocated in the shared library, it must also be freed here. (Necessary under Win32, VC++) */ void -SDL_FreeWAV(Uint8 * audio_buf) +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: */ diff -urp SDL2-2.0.8.orig/src/audio/SDL_wave.h SDL2-2.0.8/src/audio/SDL_wave.h --- SDL2-2.0.8.orig/src/audio/SDL_wave.h 2018-03-01 10:34:42.000000000 -0600 +++ SDL2-2.0.8/src/audio/SDL_wave.h 2019-08-26 10:45:44.192574222 -0500 @@ -20,11 +20,12 @@ */ #include "../SDL_internal.h" -/* WAVE files are little-endian */ +/* RIFF WAVE files are little-endian */ /*******************************************/ /* Define values for Microsoft WAVE format */ /*******************************************/ +/* FOURCC */ #define RIFF 0x46464952 /* "RIFF" */ #define WAVE 0x45564157 /* "WAVE" */ #define FACT 0x74636166 /* "fact" */ @@ -33,45 +34,116 @@ #define JUNK 0x4B4E554A /* "JUNK" */ #define FMT 0x20746D66 /* "fmt " */ #define DATA 0x61746164 /* "data" */ +/* Format tags */ +#define UNKNOWN_CODE 0x0000 #define PCM_CODE 0x0001 #define MS_ADPCM_CODE 0x0002 #define IEEE_FLOAT_CODE 0x0003 +#define ALAW_CODE 0x0006 +#define MULAW_CODE 0x0007 #define IMA_ADPCM_CODE 0x0011 -#define MP3_CODE 0x0055 +#define MPEG_CODE 0x0050 +#define MPEGLAYER3_CODE 0x0055 #define EXTENSIBLE_CODE 0xFFFE -#define WAVE_MONO 1 -#define WAVE_STEREO 2 -/* Normally, these three chunks come consecutively in a WAVE file */ -typedef struct WaveFMT +/* Stores the WAVE format information. */ +typedef struct WaveFormat { -/* Not saved in the chunk we read: - Uint32 FMTchunk; - Uint32 fmtlen; -*/ - Uint16 encoding; - Uint16 channels; /* 1 = mono, 2 = stereo */ - Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */ - Uint32 byterate; /* Average bytes per second */ - Uint16 blockalign; /* Bytes per sample block */ - Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */ -} WaveFMT; + Uint16 formattag; /* Raw value of the first field in the fmt chunk data. */ + Uint16 encoding; /* Actual encoding, possibly from the extensible header. */ + Uint16 channels; /* Number of channels. */ + Uint32 frequency; /* Sampling rate in Hz. */ + Uint32 byterate; /* Average bytes per second. */ + Uint16 blockalign; /* Bytes per block. */ + Uint16 bitspersample; /* Currently supported are 8, 16, 24, 32, and 4 for ADPCM. */ + + /* Extra information size. Number of extra bytes starting at byte 18 in the + * fmt chunk data. This is at least 22 for the extensible header. + */ + Uint16 extsize; + + /* Extensible WAVE header fields */ + Uint16 validsamplebits; + Uint32 samplesperblock; /* For compressed formats. Can be zero. Actually 16 bits in the header. */ + Uint32 channelmask; + Uint8 subformat[16]; /* A format GUID. */ +} WaveFormat; -/* The general chunk found in the WAVE file */ -typedef struct Chunk +/* Stores information on the fact chunk. */ +typedef struct WaveFact { + /* Represents the state of the fact chunk in the WAVE file. + * Set to -1 if the fact chunk is invalid. + * Set to 0 if the fact chunk is not present + * Set to 1 if the fact chunk is present and valid. + * Set to 2 if samplelength is going to be used as the number of sample frames. + */ + Sint32 status; + + /* Version 1 of the RIFF specification calls the field in the fact chunk + * dwFileSize. The Standards Update then calls it dwSampleLength and specifies + * that it is 'the length of the data in samples'. WAVE files from Windows + * with this chunk have it set to the samples per channel (sample frames). + * This is useful to truncate compressed audio to a specific sample count + * because a compressed block is usually decoded to a fixed number of + * sample frames. + */ + Uint32 samplelength; /* Raw sample length value from the fact chunk. */ +} WaveFact; + +/* Generic struct for the chunks in the WAVE file. */ +typedef struct WaveChunk { - Uint32 magic; - Uint32 length; - Uint8 *data; -} Chunk; + Uint32 fourcc; /* FOURCC of the chunk. */ + Uint32 length; /* Size of the chunk data. */ + Sint64 position; /* Position of the data in the stream. */ + Uint8 *data; /* When allocated, this points to the chunk data. length is used for the malloc size. */ + size_t size; /* Number of bytes in data that could be read from the stream. Can be smaller than length. */ +} WaveChunk; + +/* Controls how the size of the RIFF chunk affects the loading of a WAVE file. */ +typedef enum WaveRiffSizeHint { + RiffSizeNoHint, + RiffSizeChunkSearch, + RiffSizeIgnoreZero, + RiffSizeIgnore, + RiffSizeMaximum, +} WaveRiffSizeHint; + +/* Controls how a truncated WAVE file is handled. */ +typedef enum WaveTruncationHint { + TruncNoHint, + TruncVeryStrict, + TruncStrict, + TruncDropFrame, + TruncDropBlock, +} WaveTruncationHint; + +/* Controls how the fact chunk affects the loading of a WAVE file. */ +typedef enum WaveFactChunkHint { + FactNoHint, + FactTruncate, + FactStrict, + FactIgnoreZero, + FactIgnore, +} WaveFactChunkHint; -typedef struct WaveExtensibleFMT +typedef struct WaveFile { - WaveFMT format; - Uint16 size; - Uint16 validbits; - Uint32 channelmask; - Uint8 subformat[16]; /* a GUID. */ -} WaveExtensibleFMT; + WaveChunk chunk; + WaveFormat format; + WaveFact fact; + + /* Number of sample frames that will be decoded. Calculated either with the + * size of the data chunk or, if the appropriate hint is enabled, with the + * sample length value from the fact chunk. + */ + Sint64 sampleframes; + + void *decoderdata; /* Some decoders require extra data for a state. */ + + WaveRiffSizeHint riffhint; + WaveTruncationHint trunchint; + WaveFactChunkHint facthint; +} WaveFile; /* vi: set ts=4 sw=4 expandtab: */
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