File CVE-2019-13626.patch of Package SDL2.23596
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: */
