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@@ -20,675 +20,2134 @@
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*/
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#include "../SDL_internal.h"
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+#ifdef HAVE_LIMITS_H
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+#include <limits.h>
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+#else
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+#ifndef SIZE_MAX
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+#define SIZE_MAX ((size_t)-1)
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+#endif
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+#ifndef INT_MAX
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+/* Make a lucky guess. */
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+#define INT_MAX SDL_MAX_SINT32
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+#endif
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+#endif
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+
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/* Microsoft WAVE file loading routines */
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-#include "SDL_audio.h"
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-#include "SDL_wave.h"
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+#include "SDL_log.h"
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+#include "SDL_hints.h"
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+#include "SDL_audio.h"
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+#include "SDL_wave.h"
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+
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+/* Reads the value stored at the location of the f1 pointer, multiplies it
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+ * with the second argument and then stores the result to f1.
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+ * Returns 0 on success, or -1 if the multiplication overflows, in which case f1
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+ * does not get modified.
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+ */
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+static int
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+SafeMult(size_t *f1, size_t f2)
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+{
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+ if (*f1 > 0 && SIZE_MAX / *f1 <= f2) {
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+ return -1;
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+ }
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+ *f1 *= f2;
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+ return 0;
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+}
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+
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+typedef struct ADPCM_DecoderState
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+{
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+ Uint32 channels; /* Number of channels. */
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+ size_t blocksize; /* Size of an ADPCM block in bytes. */
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+ size_t blockheadersize; /* Size of an ADPCM block header in bytes. */
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+ size_t samplesperblock; /* Number of samples per channel in an ADPCM block. */
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+ size_t framesize; /* Size of a sample frame (16-bit PCM) in bytes. */
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+ Sint64 framestotal; /* Total number of sample frames. */
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+ Sint64 framesleft; /* Number of sample frames still to be decoded. */
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+ void *ddata; /* Decoder data from initialization. */
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+ void *cstate; /* Decoding state for each channel. */
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+
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+ /* ADPCM data. */
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+ struct {
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+ Uint8 *data;
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+ size_t size;
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+ size_t pos;
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+ } input;
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+
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+ /* Current ADPCM block in the ADPCM data above. */
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+ struct {
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+ Uint8 *data;
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+ size_t size;
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+ size_t pos;
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+ } block;
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+
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+ /* Decoded 16-bit PCM data. */
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+ struct {
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+ Sint16 *data;
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+ size_t size;
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+ size_t pos;
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+ } output;
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+} ADPCM_DecoderState;
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+
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+typedef struct MS_ADPCM_CoeffData
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+{
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+ Uint16 coeffcount;
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+ Sint16 *coeff;
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+ Sint16 aligndummy; /* Has to be last member. */
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+} MS_ADPCM_CoeffData;
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+
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+typedef struct MS_ADPCM_ChannelState
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+{
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+ Uint16 delta;
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+ Sint16 coeff1;
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+ Sint16 coeff2;
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+} MS_ADPCM_ChannelState;
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+
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+#ifdef SDL_WAVE_DEBUG_LOG_FORMAT
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+static void
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+WaveDebugLogFormat(WaveFile *file)
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+{
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+ WaveFormat *format = &file->format;
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+ const char *fmtstr = "WAVE file: %s, %u Hz, %s, %u bits, %u %s/s";
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+ const char *waveformat, *wavechannel, *wavebpsunit = "B";
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+ Uint32 wavebps = format->byterate;
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+ char channelstr[64];
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+
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+ SDL_zeroa(channelstr);
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+
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+ switch (format->encoding) {
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+ case PCM_CODE:
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+ waveformat = "PCM";
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+ break;
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+ case IEEE_FLOAT_CODE:
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+ waveformat = "IEEE Float";
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+ break;
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+ case ALAW_CODE:
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+ waveformat = "A-law";
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+ break;
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+ case MULAW_CODE:
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+ waveformat = "\xc2\xb5-law";
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+ break;
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+ case MS_ADPCM_CODE:
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+ waveformat = "MS ADPCM";
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+ break;
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+ case IMA_ADPCM_CODE:
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+ waveformat = "IMA ADPCM";
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+ break;
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+ default:
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+ waveformat = "Unknown";
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+ break;
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+ }
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+
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+#define SDL_WAVE_DEBUG_CHANNELCFG(STR, CODE) case CODE: wavechannel = STR; break;
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+#define SDL_WAVE_DEBUG_CHANNELSTR(STR, CODE) if (format->channelmask & CODE) { \
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+ SDL_strlcat(channelstr, channelstr[0] ? "-" STR : STR, sizeof(channelstr));}
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+
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+ if (format->formattag == EXTENSIBLE_CODE && format->channelmask > 0) {
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+ switch (format->channelmask) {
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+ SDL_WAVE_DEBUG_CHANNELCFG("1.0 Mono", 0x4)
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+ SDL_WAVE_DEBUG_CHANNELCFG("1.1 Mono", 0xc)
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+ SDL_WAVE_DEBUG_CHANNELCFG("2.0 Stereo", 0x3)
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+ SDL_WAVE_DEBUG_CHANNELCFG("2.1 Stereo", 0xb)
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+ SDL_WAVE_DEBUG_CHANNELCFG("3.0 Stereo", 0x7)
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+ SDL_WAVE_DEBUG_CHANNELCFG("3.1 Stereo", 0xf)
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+ SDL_WAVE_DEBUG_CHANNELCFG("3.0 Surround", 0x103)
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+ SDL_WAVE_DEBUG_CHANNELCFG("3.1 Surround", 0x10b)
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+ SDL_WAVE_DEBUG_CHANNELCFG("4.0 Quad", 0x33)
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+ SDL_WAVE_DEBUG_CHANNELCFG("4.1 Quad", 0x3b)
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+ SDL_WAVE_DEBUG_CHANNELCFG("4.0 Surround", 0x107)
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+ SDL_WAVE_DEBUG_CHANNELCFG("4.1 Surround", 0x10f)
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+ SDL_WAVE_DEBUG_CHANNELCFG("5.0", 0x37)
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+ SDL_WAVE_DEBUG_CHANNELCFG("5.1", 0x3f)
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+ SDL_WAVE_DEBUG_CHANNELCFG("5.0 Side", 0x607)
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+ SDL_WAVE_DEBUG_CHANNELCFG("5.1 Side", 0x60f)
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+ SDL_WAVE_DEBUG_CHANNELCFG("6.0", 0x137)
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+ SDL_WAVE_DEBUG_CHANNELCFG("6.1", 0x13f)
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+ SDL_WAVE_DEBUG_CHANNELCFG("6.0 Side", 0x707)
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+ SDL_WAVE_DEBUG_CHANNELCFG("6.1 Side", 0x70f)
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+ SDL_WAVE_DEBUG_CHANNELCFG("7.0", 0xf7)
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+ SDL_WAVE_DEBUG_CHANNELCFG("7.1", 0xff)
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+ SDL_WAVE_DEBUG_CHANNELCFG("7.0 Side", 0x6c7)
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+ SDL_WAVE_DEBUG_CHANNELCFG("7.1 Side", 0x6cf)
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+ SDL_WAVE_DEBUG_CHANNELCFG("7.0 Surround", 0x637)
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+ SDL_WAVE_DEBUG_CHANNELCFG("7.1 Surround", 0x63f)
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+ SDL_WAVE_DEBUG_CHANNELCFG("9.0 Surround", 0x5637)
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+ SDL_WAVE_DEBUG_CHANNELCFG("9.1 Surround", 0x563f)
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+ SDL_WAVE_DEBUG_CHANNELCFG("11.0 Surround", 0x56f7)
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+ SDL_WAVE_DEBUG_CHANNELCFG("11.1 Surround", 0x56ff)
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+ default:
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+ SDL_WAVE_DEBUG_CHANNELSTR("FL", 0x1)
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+ SDL_WAVE_DEBUG_CHANNELSTR("FR", 0x2)
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+ SDL_WAVE_DEBUG_CHANNELSTR("FC", 0x4)
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+ SDL_WAVE_DEBUG_CHANNELSTR("LF", 0x8)
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+ SDL_WAVE_DEBUG_CHANNELSTR("BL", 0x10)
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+ SDL_WAVE_DEBUG_CHANNELSTR("BR", 0x20)
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+ SDL_WAVE_DEBUG_CHANNELSTR("FLC", 0x40)
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+ SDL_WAVE_DEBUG_CHANNELSTR("FRC", 0x80)
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+ SDL_WAVE_DEBUG_CHANNELSTR("BC", 0x100)
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+ SDL_WAVE_DEBUG_CHANNELSTR("SL", 0x200)
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+ SDL_WAVE_DEBUG_CHANNELSTR("SR", 0x400)
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+ SDL_WAVE_DEBUG_CHANNELSTR("TC", 0x800)
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+ SDL_WAVE_DEBUG_CHANNELSTR("TFL", 0x1000)
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+ SDL_WAVE_DEBUG_CHANNELSTR("TFC", 0x2000)
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+ SDL_WAVE_DEBUG_CHANNELSTR("TFR", 0x4000)
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+ SDL_WAVE_DEBUG_CHANNELSTR("TBL", 0x8000)
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+ SDL_WAVE_DEBUG_CHANNELSTR("TBC", 0x10000)
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+ SDL_WAVE_DEBUG_CHANNELSTR("TBR", 0x20000)
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+ break;
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+ }
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+ } else {
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+ switch (format->channels) {
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+ default:
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+ if (SDL_snprintf(channelstr, sizeof(channelstr), "%u channels", format->channels) >= 0) {
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+ wavechannel = channelstr;
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+ break;
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+ }
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+ case 0:
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+ wavechannel = "Unknown";
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+ break;
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+ case 1:
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+ wavechannel = "Mono";
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+ break;
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+ case 2:
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+ wavechannel = "Setero";
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+ break;
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+ }
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+ }
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+
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+#undef SDL_WAVE_DEBUG_CHANNELCFG
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+#undef SDL_WAVE_DEBUG_CHANNELSTR
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+
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+ if (wavebps >= 1024) {
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+ wavebpsunit = "KiB";
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+ wavebps = wavebps / 1024 + (wavebps & 0x3ff ? 1 : 0);
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+ }
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+
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+ SDL_LogDebug(SDL_LOG_CATEGORY_AUDIO, fmtstr, waveformat, format->frequency, wavechannel, format->bitspersample, wavebps, wavebpsunit);
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+}
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+#endif
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+
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+#ifdef SDL_WAVE_DEBUG_DUMP_FORMAT
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+static void
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+WaveDebugDumpFormat(WaveFile *file, Uint32 rifflen, Uint32 fmtlen, Uint32 datalen)
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+{
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+ WaveFormat *format = &file->format;
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+ const char *fmtstr1 = "WAVE chunk dump:\n"
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+ "-------------------------------------------\n"
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+ "RIFF %11u\n"
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+ "-------------------------------------------\n"
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+ " fmt %11u\n"
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+ " wFormatTag 0x%04x\n"
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+ " nChannels %11u\n"
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+ " nSamplesPerSec %11u\n"
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+ " nAvgBytesPerSec %11u\n"
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+ " nBlockAlign %11u\n";
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+ const char *fmtstr2 = " wBitsPerSample %11u\n";
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+ const char *fmtstr3 = " cbSize %11u\n";
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+ const char *fmtstr4a = " wValidBitsPerSample %11u\n";
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+ const char *fmtstr4b = " wSamplesPerBlock %11u\n";
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+ const char *fmtstr5 = " dwChannelMask 0x%08x\n"
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+ " SubFormat\n"
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+ " %08x-%04x-%04x-%02x%02x%02x%02x%02x%02x%02x%02x\n";
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+ const char *fmtstr6 = "-------------------------------------------\n"
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+ " fact\n"
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+ " dwSampleLength %11u\n";
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+ const char *fmtstr7 = "-------------------------------------------\n"
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+ " data %11u\n"
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+ "-------------------------------------------\n";
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+ char *dumpstr;
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+ size_t dumppos = 0;
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+ const size_t bufsize = 1024;
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+ int res;
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+
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+ dumpstr = SDL_malloc(bufsize);
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+ if (dumpstr == NULL) {
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+ return;
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+ }
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+ dumpstr[0] = 0;
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+
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+ res = SDL_snprintf(dumpstr, bufsize, fmtstr1, rifflen, fmtlen, format->formattag, format->channels, format->frequency, format->byterate, format->blockalign);
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+ dumppos += res > 0 ? res : 0;
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+ if (fmtlen >= 16) {
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+ res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr2, format->bitspersample);
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+ dumppos += res > 0 ? res : 0;
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+ }
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+ if (fmtlen >= 18) {
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+ res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr3, format->extsize);
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+ dumppos += res > 0 ? res : 0;
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+ }
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+ if (format->formattag == EXTENSIBLE_CODE && fmtlen >= 40 && format->extsize >= 22) {
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+ const Uint8 *g = format->subformat;
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+ const Uint32 g1 = g[0] | ((Uint32)g[1] << 8) | ((Uint32)g[2] << 16) | ((Uint32)g[3] << 24);
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+ const Uint32 g2 = g[4] | ((Uint32)g[5] << 8);
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+ const Uint32 g3 = g[6] | ((Uint32)g[7] << 8);
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+
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+ switch (format->encoding) {
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+ default:
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+ res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr4a, format->validsamplebits);
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+ dumppos += res > 0 ? res : 0;
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+ break;
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+ case MS_ADPCM_CODE:
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+ case IMA_ADPCM_CODE:
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+ res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr4b, format->samplesperblock);
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+ dumppos += res > 0 ? res : 0;
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+ break;
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+ }
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+ 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]);
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+ dumppos += res > 0 ? res : 0;
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+ } else {
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+ switch (format->encoding) {
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+ case MS_ADPCM_CODE:
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+ case IMA_ADPCM_CODE:
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+ if (fmtlen >= 20 && format->extsize >= 2) {
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+ res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr4b, format->samplesperblock);
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+ dumppos += res > 0 ? res : 0;
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+ }
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+ break;
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+ }
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+ }
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+ if (file->fact.status >= 1) {
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+ res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr6, file->fact.samplelength);
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+ dumppos += res > 0 ? res : 0;
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+ }
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+ res = SDL_snprintf(dumpstr + dumppos, bufsize - dumppos, fmtstr7, datalen);
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+ dumppos += res > 0 ? res : 0;
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+
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+ SDL_LogDebug(SDL_LOG_CATEGORY_AUDIO, "%s", dumpstr);
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+
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+ free(dumpstr);
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+}
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+#endif
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+
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+static Sint64
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+WaveAdjustToFactValue(WaveFile *file, Sint64 sampleframes)
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+{
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+ if (file->fact.status == 2) {
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+ if (file->facthint == FactStrict && sampleframes < file->fact.samplelength) {
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+ return SDL_SetError("Invalid number of sample frames in WAVE fact chunk (too many)");
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+ } else if (sampleframes > file->fact.samplelength) {
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+ return file->fact.samplelength;
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+ }
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+ }
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+
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+ return sampleframes;
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+}
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+
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+static int
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+MS_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
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+{
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+ WaveFormat *format = &file->format;
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+ const size_t blockheadersize = (size_t)file->format.channels * 7;
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+ const size_t availableblocks = datalength / file->format.blockalign;
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+ const size_t blockframebitsize = (size_t)file->format.bitspersample * file->format.channels;
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+ const size_t trailingdata = datalength % file->format.blockalign;
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+
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+ if (file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict) {
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+ /* The size of the data chunk must be a multiple of the block size. */
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+ if (datalength < blockheadersize || trailingdata > 0) {
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+ return SDL_SetError("Truncated MS ADPCM block");
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+ }
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+ }
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+
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+ /* Calculate number of sample frames that will be decoded. */
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+ file->sampleframes = (Sint64)availableblocks * format->samplesperblock;
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+ if (trailingdata > 0) {
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+ /* The last block is truncated. Check if we can get any samples out of it. */
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+ if (file->trunchint == TruncDropFrame) {
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+ /* Drop incomplete sample frame. */
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+ if (trailingdata >= blockheadersize) {
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+ size_t trailingsamples = 2 + (trailingdata - blockheadersize) * 8 / blockframebitsize;
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+ if (trailingsamples > format->samplesperblock) {
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+ trailingsamples = format->samplesperblock;
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+ }
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+ file->sampleframes += trailingsamples;
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+ }
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+ }
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+ }
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+
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+ file->sampleframes = WaveAdjustToFactValue(file, file->sampleframes);
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+ if (file->sampleframes < 0) {
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+ return -1;
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+ }
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+
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+ return 0;
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+}
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+
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+static int
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+MS_ADPCM_Init(WaveFile *file, size_t datalength)
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+{
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+ WaveFormat *format = &file->format;
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+ WaveChunk *chunk = &file->chunk;
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+ const size_t blockheadersize = (size_t)format->channels * 7;
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+ const size_t blockdatasize = (size_t)format->blockalign - blockheadersize;
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+ const size_t blockframebitsize = (size_t)format->bitspersample * format->channels;
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+ const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize;
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|
|
+ const Sint16 presetcoeffs[14] = {256, 0, 512, -256, 0, 0, 192, 64, 240, 0, 460, -208, 392, -232};
|
|
|
+ size_t i, coeffcount;
|
|
|
+ MS_ADPCM_CoeffData *coeffdata;
|
|
|
+
|
|
|
+ /* Sanity checks. */
|
|
|
+
|
|
|
+ /* While it's clear how IMA ADPCM handles more than two channels, the nibble
|
|
|
+ * order of MS ADPCM makes it awkward. The Standards Update does not talk
|
|
|
+ * about supporting more than stereo anyway.
|
|
|
+ */
|
|
|
+ if (format->channels > 2) {
|
|
|
+ return SDL_SetError("Invalid number of channels");
|
|
|
+ }
|
|
|
+
|
|
|
+ if (format->bitspersample != 4) {
|
|
|
+ return SDL_SetError("Invalid MS ADPCM bits per sample of %u", (unsigned int)format->bitspersample);
|
|
|
+ }
|
|
|
+
|
|
|
+ /* The block size must be big enough to contain the block header. */
|
|
|
+ if (format->blockalign < blockheadersize) {
|
|
|
+ return SDL_SetError("Invalid MS ADPCM block size (nBlockAlign)");
|
|
|
+ }
|
|
|
+
|
|
|
+ if (format->formattag == EXTENSIBLE_CODE) {
|
|
|
+ /* Does have a GUID (like all format tags), but there's no specification
|
|
|
+ * for how the data is packed into the extensible header. Making
|
|
|
+ * assumptions here could lead to new formats nobody wants to support.
|
|
|
+ */
|
|
|
+ return SDL_SetError("MS ADPCM with the extensible header is not supported");
|
|
|
+ }
|
|
|
+
|
|
|
+ /* There are wSamplesPerBlock, wNumCoef, and at least 7 coefficient pairs in
|
|
|
+ * the extended part of the header.
|
|
|
+ */
|
|
|
+ if (chunk->size < 22) {
|
|
|
+ return SDL_SetError("Could not read MS ADPCM format header");
|
|
|
+ }
|
|
|
+
|
|
|
+ format->samplesperblock = chunk->data[18] | ((Uint16)chunk->data[19] << 8);
|
|
|
+ /* Number of coefficient pairs. A pair has two 16-bit integers. */
|
|
|
+ coeffcount = chunk->data[20] | ((size_t)chunk->data[21] << 8);
|
|
|
+ /* bPredictor, the integer offset into the coefficients array, is only
|
|
|
+ * 8 bits. It can only address the first 256 coefficients. Let's limit
|
|
|
+ * the count number here.
|
|
|
+ */
|
|
|
+ if (coeffcount > 256) {
|
|
|
+ coeffcount = 256;
|
|
|
+ }
|
|
|
+
|
|
|
+ if (chunk->size < 22 + coeffcount * 4) {
|
|
|
+ return SDL_SetError("Could not read custom coefficients in MS ADPCM format header");
|
|
|
+ } else if (format->extsize < 4 + coeffcount * 4) {
|
|
|
+ return SDL_SetError("Invalid MS ADPCM format header (too small)");
|
|
|
+ } else if (coeffcount < 7) {
|
|
|
+ return SDL_SetError("Missing required coefficients in MS ADPCM format header");
|
|
|
+ }
|
|
|
+
|
|
|
+ coeffdata = (MS_ADPCM_CoeffData *)SDL_malloc(sizeof(MS_ADPCM_CoeffData) + coeffcount * 4);
|
|
|
+ file->decoderdata = coeffdata; /* Freed in cleanup. */
|
|
|
+ if (coeffdata == NULL) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ }
|
|
|
+ coeffdata->coeff = &coeffdata->aligndummy;
|
|
|
+ coeffdata->coeffcount = (Uint16)coeffcount;
|
|
|
+
|
|
|
+ /* Copy the 16-bit pairs. */
|
|
|
+ for (i = 0; i < coeffcount * 2; i++) {
|
|
|
+ Sint32 c = chunk->data[22 + i * 2] | ((Sint32)chunk->data[23 + i * 2] << 8);
|
|
|
+ if (c >= 0x8000) {
|
|
|
+ c -= 0x10000;
|
|
|
+ }
|
|
|
+ if (i < 14 && c != presetcoeffs[i]) {
|
|
|
+ return SDL_SetError("Wrong preset coefficients in MS ADPCM format header");
|
|
|
+ }
|
|
|
+ coeffdata->coeff[i] = (Sint16)c;
|
|
|
+ }
|
|
|
+
|
|
|
+ /* Technically, wSamplesPerBlock is required, but we have all the
|
|
|
+ * information in the other fields to calculate it, if it's zero.
|
|
|
+ */
|
|
|
+ if (format->samplesperblock == 0) {
|
|
|
+ /* Let's be nice to the encoders that didn't know how to fill this.
|
|
|
+ * The Standards Update calculates it this way:
|
|
|
+ *
|
|
|
+ * x = Block size (in bits) minus header size (in bits)
|
|
|
+ * y = Bit depth multiplied by channel count
|
|
|
+ * z = Number of samples per channel in block header
|
|
|
+ * wSamplesPerBlock = x / y + z
|
|
|
+ */
|
|
|
+ format->samplesperblock = (Uint32)blockdatasamples + 2;
|
|
|
+ }
|
|
|
+
|
|
|
+ /* nBlockAlign can be in conflict with wSamplesPerBlock. For example, if
|
|
|
+ * the number of samples doesn't fit into the block. The Standards Update
|
|
|
+ * also describes wSamplesPerBlock with a formula that makes it necessary to
|
|
|
+ * always fill the block with the maximum amount of samples, but this is not
|
|
|
+ * enforced here as there are no compatibility issues.
|
|
|
+ * A truncated block header with just one sample is not supported.
|
|
|
+ */
|
|
|
+ if (format->samplesperblock == 1 || blockdatasamples < format->samplesperblock - 2) {
|
|
|
+ return SDL_SetError("Invalid number of samples per MS ADPCM block (wSamplesPerBlock)");
|
|
|
+ }
|
|
|
+
|
|
|
+ if (MS_ADPCM_CalculateSampleFrames(file, datalength) < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static Sint16
|
|
|
+MS_ADPCM_ProcessNibble(MS_ADPCM_ChannelState *cstate, Sint32 sample1, Sint32 sample2, Uint8 nybble)
|
|
|
+{
|
|
|
+ const Sint32 max_audioval = 32767;
|
|
|
+ const Sint32 min_audioval = -32768;
|
|
|
+ const Uint16 max_deltaval = 65535;
|
|
|
+ const Uint16 adaptive[] = {
|
|
|
+ 230, 230, 230, 230, 307, 409, 512, 614,
|
|
|
+ 768, 614, 512, 409, 307, 230, 230, 230
|
|
|
+ };
|
|
|
+ Sint32 new_sample;
|
|
|
+ Sint32 errordelta;
|
|
|
+ Uint32 delta = cstate->delta;
|
|
|
+
|
|
|
+ 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 = (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;
|
|
|
+ }
|
|
|
+
|
|
|
+ cstate->delta = (Uint16)delta;
|
|
|
+ return (Sint16)new_sample;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+MS_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
|
|
|
+{
|
|
|
+ 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);
|
|
|
+
|
|
|
+ /* Load the samples from the header. Interestingly, the sample later in
|
|
|
+ * the output stream comes first.
|
|
|
+ */
|
|
|
+ o = channels * 3 + c * 2;
|
|
|
+ sample = state->block.data[o] | ((Sint32)state->block.data[o + 1] << 8);
|
|
|
+ if (sample >= 0x8000) {
|
|
|
+ sample -= 0x10000;
|
|
|
+ }
|
|
|
+ state->output.data[state->output.pos + channels] = (Sint16)sample;
|
|
|
+
|
|
|
+ o = channels * 5 + c * 2;
|
|
|
+ sample = state->block.data[o] | ((Sint32)state->block.data[o + 1] << 8);
|
|
|
+ if (sample >= 0x8000) {
|
|
|
+ sample -= 0x10000;
|
|
|
+ }
|
|
|
+ state->output.data[state->output.pos] = (Sint16)sample;
|
|
|
+
|
|
|
+ state->output.pos++;
|
|
|
+ }
|
|
|
+
|
|
|
+ state->block.pos += state->blockheadersize;
|
|
|
+
|
|
|
+ /* Skip second sample frame that came from the header. */
|
|
|
+ state->output.pos += state->channels;
|
|
|
+
|
|
|
+ /* Header provided two sample frames. */
|
|
|
+ state->framesleft -= 2;
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+/* Decodes the data of the MS ADPCM block. Decoding will stop if a block is too
|
|
|
+ * short, returning with none or partially decoded data. The partial data
|
|
|
+ * will always contain full sample frames (same sample count for each channel).
|
|
|
+ * Incomplete sample frames are discarded.
|
|
|
+ */
|
|
|
+static int
|
|
|
+MS_ADPCM_DecodeBlockData(ADPCM_DecoderState *state)
|
|
|
+{
|
|
|
+ Uint16 nybble = 0;
|
|
|
+ Sint16 sample1, sample2;
|
|
|
+ const Uint32 channels = state->channels;
|
|
|
+ Uint32 c;
|
|
|
+ MS_ADPCM_ChannelState *cstate = (MS_ADPCM_ChannelState *)state->cstate;
|
|
|
+
|
|
|
+ size_t blockpos = state->block.pos;
|
|
|
+ size_t blocksize = state->block.size;
|
|
|
+
|
|
|
+ size_t outpos = state->output.pos;
|
|
|
+
|
|
|
+ Sint64 blockframesleft = state->samplesperblock - 2;
|
|
|
+ if (blockframesleft > state->framesleft) {
|
|
|
+ blockframesleft = state->framesleft;
|
|
|
+ }
|
|
|
+
|
|
|
+ while (blockframesleft > 0) {
|
|
|
+ for (c = 0; c < channels; c++) {
|
|
|
+ if (nybble & 0x4000) {
|
|
|
+ nybble <<= 4;
|
|
|
+ } else if (blockpos < blocksize) {
|
|
|
+ nybble = state->block.data[blockpos++] | 0x4000;
|
|
|
+ } else {
|
|
|
+ /* Out of input data. Drop the incomplete frame and return. */
|
|
|
+ state->output.pos = outpos - c;
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+
|
|
|
+ /* Load previous samples which may come from the block header. */
|
|
|
+ sample1 = state->output.data[outpos - channels];
|
|
|
+ sample2 = state->output.data[outpos - channels * 2];
|
|
|
+
|
|
|
+ sample1 = MS_ADPCM_ProcessNibble(cstate + c, sample1, sample2, (nybble >> 4) & 0x0f);
|
|
|
+ state->output.data[outpos++] = sample1;
|
|
|
+ }
|
|
|
+
|
|
|
+ state->framesleft--;
|
|
|
+ blockframesleft--;
|
|
|
+ }
|
|
|
+
|
|
|
+ state->output.pos = outpos;
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+MS_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
|
|
|
+{
|
|
|
+ int result;
|
|
|
+ size_t bytesleft, outputsize;
|
|
|
+ WaveChunk *chunk = &file->chunk;
|
|
|
+ ADPCM_DecoderState state;
|
|
|
+ MS_ADPCM_ChannelState cstate[2];
|
|
|
+
|
|
|
+ SDL_zero(state);
|
|
|
+ SDL_zeroa(cstate);
|
|
|
+
|
|
|
+ if (chunk->size != chunk->length) {
|
|
|
+ /* Could not read everything. Recalculate number of sample frames. */
|
|
|
+ if (MS_ADPCM_CalculateSampleFrames(file, chunk->size) < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ /* Nothing to decode, nothing to return. */
|
|
|
+ if (file->sampleframes == 0) {
|
|
|
+ *audio_buf = NULL;
|
|
|
+ *audio_len = 0;
|
|
|
+ return 0;
|
|
|
+ }
|
|
|
+
|
|
|
+ state.blocksize = file->format.blockalign;
|
|
|
+ state.channels = file->format.channels;
|
|
|
+ state.blockheadersize = (size_t)state.channels * 7;
|
|
|
+ state.samplesperblock = file->format.samplesperblock;
|
|
|
+ state.framesize = state.channels * sizeof(Sint16);
|
|
|
+ state.ddata = file->decoderdata;
|
|
|
+ state.framestotal = file->sampleframes;
|
|
|
+ state.framesleft = state.framestotal;
|
|
|
+
|
|
|
+ state.input.data = chunk->data;
|
|
|
+ state.input.size = chunk->size;
|
|
|
+ state.input.pos = 0;
|
|
|
+
|
|
|
+ /* The output size in bytes. May get modified if data is truncated. */
|
|
|
+ outputsize = (size_t)state.framestotal;
|
|
|
+ if (SafeMult(&outputsize, state.framesize)) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ } else if (outputsize > SDL_MAX_UINT32 || state.framestotal > SIZE_MAX) {
|
|
|
+ return SDL_SetError("WAVE file too big");
|
|
|
+ }
|
|
|
+
|
|
|
+ state.output.pos = 0;
|
|
|
+ state.output.size = outputsize / sizeof(Sint16);
|
|
|
+ state.output.data = (Sint16 *)SDL_malloc(outputsize);
|
|
|
+ if (state.output.data == NULL) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ }
|
|
|
+
|
|
|
+ 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 == TruncStrict) {
|
|
|
+ SDL_free(state.output.data);
|
|
|
+ return SDL_SetError("Truncated data chunk");
|
|
|
+ } else if (file->trunchint != TruncDropFrame) {
|
|
|
+ state.output.pos -= state.output.pos % (state.samplesperblock * state.channels);
|
|
|
+ }
|
|
|
+ outputsize = state.output.pos * sizeof(Sint16); /* Can't overflow, is always smaller. */
|
|
|
+ break;
|
|
|
+ }
|
|
|
+
|
|
|
+ state.input.pos += state.block.size;
|
|
|
+ bytesleft = state.input.size - state.input.pos;
|
|
|
+ }
|
|
|
+
|
|
|
+ *audio_buf = (Uint8 *)state.output.data;
|
|
|
+ *audio_len = (Uint32)outputsize;
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+IMA_ADPCM_CalculateSampleFrames(WaveFile *file, size_t datalength)
|
|
|
+{
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
+ const size_t blockheadersize = (size_t)format->channels * 4;
|
|
|
+ const size_t subblockframesize = (size_t)format->channels * 4;
|
|
|
+ const size_t availableblocks = datalength / format->blockalign;
|
|
|
+ const size_t trailingdata = datalength % format->blockalign;
|
|
|
+
|
|
|
+ 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 = (size_t)format->channels * 4;
|
|
|
+ const size_t blockdatasize = (size_t)format->blockalign - blockheadersize;
|
|
|
+ const size_t blockframebitsize = (size_t)format->bitspersample * format->channels;
|
|
|
+ const size_t blockdatasamples = (blockdatasize * 8) / blockframebitsize;
|
|
|
+
|
|
|
+ /* Sanity checks. */
|
|
|
+
|
|
|
+ /* IMA ADPCM can also have 3-bit samples, but it's not supported by SDL at this time. */
|
|
|
+ if (format->bitspersample == 3) {
|
|
|
+ return SDL_SetError("3-bit IMA ADPCM currently not supported");
|
|
|
+ } else if (format->bitspersample != 4) {
|
|
|
+ return SDL_SetError("Invalid IMA ADPCM bits per sample of %u", (unsigned int)format->bitspersample);
|
|
|
+ }
|
|
|
+
|
|
|
+ /* The block size is required to be a multiple of 4 and it must be able to
|
|
|
+ * hold a block header.
|
|
|
+ */
|
|
|
+ if (format->blockalign < blockheadersize || format->blockalign % 4) {
|
|
|
+ return SDL_SetError("Invalid IMA ADPCM block size (nBlockAlign)");
|
|
|
+ }
|
|
|
+
|
|
|
+ if (format->formattag == EXTENSIBLE_CODE) {
|
|
|
+ /* There's no specification for this, but it's basically the same
|
|
|
+ * format because the extensible header has wSampePerBlocks too.
|
|
|
+ */
|
|
|
+ } else {
|
|
|
+ /* The Standards Update says there 'should' be 2 bytes for wSamplesPerBlock. */
|
|
|
+ if (chunk->size >= 20 && format->extsize >= 2) {
|
|
|
+ format->samplesperblock = chunk->data[18] | ((Uint16)chunk->data[19] << 8);
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ if (format->samplesperblock == 0) {
|
|
|
+ /* Field zero? No problem. We just assume the encoder packed the block.
|
|
|
+ * The specification calculates it this way:
|
|
|
+ *
|
|
|
+ * x = Block size (in bits) minus header size (in bits)
|
|
|
+ * y = Bit depth multiplied by channel count
|
|
|
+ * z = Number of samples per channel in header
|
|
|
+ * wSamplesPerBlock = x / y + z
|
|
|
+ */
|
|
|
+ format->samplesperblock = (Uint32)blockdatasamples + 1;
|
|
|
+ }
|
|
|
+
|
|
|
+ /* nBlockAlign can be in conflict with wSamplesPerBlock. For example, if
|
|
|
+ * the number of samples doesn't fit into the block. The Standards Update
|
|
|
+ * also describes wSamplesPerBlock with a formula that makes it necessary
|
|
|
+ * to always fill the block with the maximum amount of samples, but this is
|
|
|
+ * not enforced here as there are no compatibility issues.
|
|
|
+ */
|
|
|
+ if (blockdatasamples < format->samplesperblock - 1) {
|
|
|
+ return SDL_SetError("Invalid number of samples per IMA ADPCM block (wSamplesPerBlock)");
|
|
|
+ }
|
|
|
+
|
|
|
+ if (IMA_ADPCM_CalculateSampleFrames(file, datalength) < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static Sint16
|
|
|
+IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 nybble)
|
|
|
+{
|
|
|
+ const Sint32 max_audioval = 32767;
|
|
|
+ const Sint32 min_audioval = -32768;
|
|
|
+ const Sint8 index_table_4b[16] = {
|
|
|
+ -1, -1, -1, -1,
|
|
|
+ 2, 4, 6, 8,
|
|
|
+ -1, -1, -1, -1,
|
|
|
+ 2, 4, 6, 8
|
|
|
+ };
|
|
|
+ const 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,
|
|
|
+ 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,
|
|
|
+ 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
|
|
|
+ 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630,
|
|
|
+ 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,
|
|
|
+ 22385, 24623, 27086, 29794, 32767
|
|
|
+ };
|
|
|
+ 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[(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;
|
|
|
+ if (nybble & 0x01)
|
|
|
+ delta += step >> 2;
|
|
|
+ if (nybble & 0x08)
|
|
|
+ delta = -delta;
|
|
|
+
|
|
|
+ sample = lastsample + delta;
|
|
|
+
|
|
|
+ /* Clamp output sample */
|
|
|
+ if (sample > max_audioval) {
|
|
|
+ sample = max_audioval;
|
|
|
+ } else if (sample < min_audioval) {
|
|
|
+ sample = min_audioval;
|
|
|
+ }
|
|
|
+
|
|
|
+ return (Sint16)sample;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+IMA_ADPCM_DecodeBlockHeader(ADPCM_DecoderState *state)
|
|
|
+{
|
|
|
+ Sint16 step;
|
|
|
+ Uint32 c;
|
|
|
+ Uint8 *cstate = state->cstate;
|
|
|
+
|
|
|
+ for (c = 0; c < state->channels; c++) {
|
|
|
+ size_t o = state->block.pos + c * 4;
|
|
|
+
|
|
|
+ /* 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;
|
|
|
+
|
|
|
+ /* 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_DecodeBlockData(ADPCM_DecoderState *state)
|
|
|
+{
|
|
|
+ size_t i;
|
|
|
+ int retval = 0;
|
|
|
+ const Uint32 channels = state->channels;
|
|
|
+ const size_t subblockframesize = channels * 4;
|
|
|
+ Uint64 bytesrequired;
|
|
|
+ Uint32 c;
|
|
|
+
|
|
|
+ size_t blockpos = state->block.pos;
|
|
|
+ size_t blocksize = state->block.size;
|
|
|
+ size_t blockleft = blocksize - blockpos;
|
|
|
+
|
|
|
+ size_t outpos = state->output.pos;
|
|
|
+
|
|
|
+ Sint64 blockframesleft = state->samplesperblock - 1;
|
|
|
+ if (blockframesleft > state->framesleft) {
|
|
|
+ blockframesleft = state->framesleft;
|
|
|
+ }
|
|
|
+
|
|
|
+ bytesrequired = (blockframesleft + 7) / 8 * subblockframesize;
|
|
|
+ if (blockleft < bytesrequired) {
|
|
|
+ /* Data truncated. Calculate how many samples we can get out if it. */
|
|
|
+ const size_t guaranteedframes = blockleft / subblockframesize;
|
|
|
+ const size_t remainingbytes = blockleft % subblockframesize;
|
|
|
+ blockframesleft = guaranteedframes;
|
|
|
+ if (remainingbytes > subblockframesize - 4) {
|
|
|
+ blockframesleft += (remainingbytes % 4) * 2;
|
|
|
+ }
|
|
|
+ /* Signal the truncation. */
|
|
|
+ retval = -1;
|
|
|
+ }
|
|
|
+
|
|
|
+ /* Each channel has their nibbles packed into 32-bit blocks. These blocks
|
|
|
+ * are interleaved and make up the data part of the ADPCM block. This loop
|
|
|
+ * decodes the samples as they come from the input data and puts them at
|
|
|
+ * the appropriate places in the output data.
|
|
|
+ */
|
|
|
+ while (blockframesleft > 0) {
|
|
|
+ const size_t subblocksamples = blockframesleft < 8 ? (size_t)blockframesleft : 8;
|
|
|
+
|
|
|
+ for (c = 0; c < channels; c++) {
|
|
|
+ Uint8 nybble = 0;
|
|
|
+ /* Load previous sample which may come from the block header. */
|
|
|
+ Sint16 sample = state->output.data[outpos + c - channels];
|
|
|
+
|
|
|
+ for (i = 0; i < subblocksamples; i++) {
|
|
|
+ if (i & 1) {
|
|
|
+ nybble >>= 4;
|
|
|
+ } else {
|
|
|
+ nybble = state->block.data[blockpos++];
|
|
|
+ }
|
|
|
+
|
|
|
+ sample = IMA_ADPCM_ProcessNibble((Sint8 *)state->cstate + c, sample, nybble & 0x0f);
|
|
|
+ state->output.data[outpos + c + i * channels] = sample;
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ outpos += channels * subblocksamples;
|
|
|
+ state->framesleft -= subblocksamples;
|
|
|
+ blockframesleft -= subblocksamples;
|
|
|
+ }
|
|
|
+
|
|
|
+ state->block.pos = blockpos;
|
|
|
+ state->output.pos = outpos;
|
|
|
+
|
|
|
+ return retval;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+IMA_ADPCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
|
|
|
+{
|
|
|
+ int result;
|
|
|
+ size_t bytesleft, outputsize;
|
|
|
+ WaveChunk *chunk = &file->chunk;
|
|
|
+ ADPCM_DecoderState state;
|
|
|
+ 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;
|
|
|
+ }
|
|
|
+
|
|
|
+ SDL_zero(state);
|
|
|
+ state.channels = file->format.channels;
|
|
|
+ state.blocksize = file->format.blockalign;
|
|
|
+ state.blockheadersize = (size_t)state.channels * 4;
|
|
|
+ state.samplesperblock = file->format.samplesperblock;
|
|
|
+ state.framesize = state.channels * sizeof(Sint16);
|
|
|
+ state.framestotal = file->sampleframes;
|
|
|
+ state.framesleft = state.framestotal;
|
|
|
+
|
|
|
+ state.input.data = chunk->data;
|
|
|
+ state.input.size = chunk->size;
|
|
|
+ state.input.pos = 0;
|
|
|
+
|
|
|
+ /* The output size in bytes. May get modified if data is truncated. */
|
|
|
+ outputsize = (size_t)state.framestotal;
|
|
|
+ if (SafeMult(&outputsize, state.framesize)) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ } else if (outputsize > SDL_MAX_UINT32 || state.framestotal > SIZE_MAX) {
|
|
|
+ return SDL_SetError("WAVE file too big");
|
|
|
+ }
|
|
|
+
|
|
|
+ state.output.pos = 0;
|
|
|
+ state.output.size = outputsize / sizeof(Sint16);
|
|
|
+ state.output.data = (Sint16 *)SDL_malloc(outputsize);
|
|
|
+ if (state.output.data == NULL) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ }
|
|
|
+
|
|
|
+ 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);
|
|
|
+ if (result == 0) {
|
|
|
+ /* Decode the block data. It stores the samples directly in the output. */
|
|
|
+ result = IMA_ADPCM_DecodeBlockData(&state);
|
|
|
+ }
|
|
|
+
|
|
|
+ if (result == -1) {
|
|
|
+ /* Unexpected end. Stop decoding and return partial data if necessary. */
|
|
|
+ if (file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict) {
|
|
|
+ SDL_free(state.output.data);
|
|
|
+ SDL_free(cstate);
|
|
|
+ return SDL_SetError("Truncated data chunk");
|
|
|
+ } else if (file->trunchint != TruncDropFrame) {
|
|
|
+ state.output.pos -= state.output.pos % (state.samplesperblock * state.channels);
|
|
|
+ }
|
|
|
+ outputsize = state.output.pos * sizeof(Sint16); /* Can't overflow, is always smaller. */
|
|
|
+ break;
|
|
|
+ }
|
|
|
+
|
|
|
+ state.input.pos += state.block.size;
|
|
|
+ bytesleft = state.input.size - state.input.pos;
|
|
|
+ }
|
|
|
+
|
|
|
+ *audio_buf = (Uint8 *)state.output.data;
|
|
|
+ *audio_len = (Uint32)outputsize;
|
|
|
+
|
|
|
+ SDL_free(cstate);
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+LAW_Init(WaveFile *file, size_t datalength)
|
|
|
+{
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
+
|
|
|
+ /* Standards Update requires this to be 8. */
|
|
|
+ if (format->bitspersample != 8) {
|
|
|
+ return SDL_SetError("Invalid companded bits per sample of %u", (unsigned int)format->bitspersample);
|
|
|
+ }
|
|
|
+
|
|
|
+ /* Not going to bother with weird padding. */
|
|
|
+ if (format->blockalign != format->channels) {
|
|
|
+ return SDL_SetError("Unsupported block alignment");
|
|
|
+ }
|
|
|
+
|
|
|
+ if ((file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict)) {
|
|
|
+ if (format->blockalign > 1 && datalength % format->blockalign) {
|
|
|
+ return SDL_SetError("Truncated data chunk in WAVE file");
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ file->sampleframes = WaveAdjustToFactValue(file, datalength / format->blockalign);
|
|
|
+ if (file->sampleframes < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+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 (SafeMult(&sample_count, format->channels)) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ }
|
|
|
+
|
|
|
+ expanded_len = sample_count;
|
|
|
+ if (SafeMult(&expanded_len, sizeof(Sint16))) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ } else if (expanded_len > SDL_MAX_UINT32 || file->sampleframes > SIZE_MAX) {
|
|
|
+ return SDL_SetError("WAVE file too big");
|
|
|
+ }
|
|
|
+
|
|
|
+ /* 1 to avoid allocating zero bytes, to keep static analysis happy. */
|
|
|
+ src = (Uint8 *)SDL_realloc(chunk->data, expanded_len ? expanded_len : 1);
|
|
|
+ if (src == NULL) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ }
|
|
|
+ chunk->data = NULL;
|
|
|
+ chunk->size = 0;
|
|
|
+
|
|
|
+ dst = (Sint16 *)src;
|
|
|
+
|
|
|
+ /* 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;
|
|
|
+ }
|
|
|
+ mantissa = (mantissa << 4) | 0x8;
|
|
|
+ if (exponent > 1) {
|
|
|
+ mantissa <<= exponent - 1;
|
|
|
+ }
|
|
|
+
|
|
|
+ dst[i] = nibble & 0x80 ? mantissa : -mantissa;
|
|
|
+ }
|
|
|
+ break;
|
|
|
+ case MULAW_CODE:
|
|
|
+ while (i--) {
|
|
|
+ Uint8 nibble = ~src[i];
|
|
|
+ Sint16 mantissa = nibble & 0xf;
|
|
|
+ Uint8 exponent = (nibble >> 4) & 0x7;
|
|
|
+ Sint16 step = 4 << (exponent + 1);
|
|
|
+
|
|
|
+ mantissa = (0x80 << exponent) + step * mantissa + step / 2 - 132;
|
|
|
+
|
|
|
+ dst[i] = nibble & 0x80 ? -mantissa : mantissa;
|
|
|
+ }
|
|
|
+ break;
|
|
|
+#endif
|
|
|
+ default:
|
|
|
+ SDL_free(src);
|
|
|
+ return SDL_SetError("Unknown companded encoding");
|
|
|
+ }
|
|
|
+
|
|
|
+ *audio_buf = src;
|
|
|
+ *audio_len = (Uint32)expanded_len;
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+PCM_Init(WaveFile *file, size_t datalength)
|
|
|
+{
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
+
|
|
|
+ if (format->encoding == PCM_CODE) {
|
|
|
+ switch (format->bitspersample) {
|
|
|
+ case 8:
|
|
|
+ case 16:
|
|
|
+ case 24:
|
|
|
+ case 32:
|
|
|
+ /* These are supported. */
|
|
|
+ break;
|
|
|
+ default:
|
|
|
+ return SDL_SetError("%u-bit PCM format not supported", (unsigned int)format->bitspersample);
|
|
|
+ }
|
|
|
+ } else if (format->encoding == IEEE_FLOAT_CODE) {
|
|
|
+ if (format->bitspersample != 32) {
|
|
|
+ return SDL_SetError("%u-bit IEEE floating-point format not supported", (unsigned int)format->bitspersample);
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ /* It wouldn't be that hard to support more exotic block sizes, but
|
|
|
+ * the most common formats should do for now.
|
|
|
+ */
|
|
|
+ if (format->blockalign * 8 != format->channels * format->bitspersample) {
|
|
|
+ return SDL_SetError("Unsupported block alignment");
|
|
|
+ }
|
|
|
+
|
|
|
+ if ((file->trunchint == TruncVeryStrict || file->trunchint == TruncStrict)) {
|
|
|
+ if (format->blockalign > 1 && datalength % format->blockalign) {
|
|
|
+ return SDL_SetError("Truncated data chunk in WAVE file");
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ file->sampleframes = WaveAdjustToFactValue(file, datalength / format->blockalign);
|
|
|
+ if (file->sampleframes < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+PCM_ConvertSint24ToSint32(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
|
|
|
+{
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
+ WaveChunk *chunk = &file->chunk;
|
|
|
+ size_t i, expanded_len, sample_count;
|
|
|
+ Uint8 *ptr;
|
|
|
+
|
|
|
+ sample_count = (size_t)file->sampleframes;
|
|
|
+ if (SafeMult(&sample_count, format->channels)) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ }
|
|
|
+
|
|
|
+ expanded_len = sample_count;
|
|
|
+ if (SafeMult(&expanded_len, sizeof(Sint32))) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ } else if (expanded_len > SDL_MAX_UINT32 || file->sampleframes > SIZE_MAX) {
|
|
|
+ return SDL_SetError("WAVE file too big");
|
|
|
+ }
|
|
|
+
|
|
|
+ /* 1 to avoid allocating zero bytes, to keep static analysis happy. */
|
|
|
+ ptr = (Uint8 *)SDL_realloc(chunk->data, expanded_len ? expanded_len : 1);
|
|
|
+ if (ptr == NULL) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ }
|
|
|
+
|
|
|
+ /* This pointer is now invalid. */
|
|
|
+ chunk->data = NULL;
|
|
|
+ chunk->size = 0;
|
|
|
+
|
|
|
+ *audio_buf = ptr;
|
|
|
+ *audio_len = (Uint32)expanded_len;
|
|
|
+
|
|
|
+ /* work from end to start, since we're expanding in-place. */
|
|
|
+ for (i = sample_count; i > 0; i--) {
|
|
|
+ const size_t o = i - 1;
|
|
|
+ uint8_t b[4];
|
|
|
+
|
|
|
+ b[0] = 0;
|
|
|
+ b[1] = ptr[o * 3];
|
|
|
+ b[2] = ptr[o * 3 + 1];
|
|
|
+ b[3] = ptr[o * 3 + 2];
|
|
|
+
|
|
|
+ ptr[o * 4 + 0] = b[0];
|
|
|
+ ptr[o * 4 + 1] = b[1];
|
|
|
+ ptr[o * 4 + 2] = b[2];
|
|
|
+ ptr[o * 4 + 3] = b[3];
|
|
|
+ }
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+PCM_Decode(WaveFile *file, Uint8 **audio_buf, Uint32 *audio_len)
|
|
|
+{
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
+ WaveChunk *chunk = &file->chunk;
|
|
|
+ size_t outputsize;
|
|
|
+
|
|
|
+ if (chunk->length != chunk->size) {
|
|
|
+ file->sampleframes = WaveAdjustToFactValue(file, chunk->size / format->blockalign);
|
|
|
+ if (file->sampleframes < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ /* Nothing to decode, nothing to return. */
|
|
|
+ if (file->sampleframes == 0) {
|
|
|
+ *audio_buf = NULL;
|
|
|
+ *audio_len = 0;
|
|
|
+ return 0;
|
|
|
+ }
|
|
|
+
|
|
|
+ /* 24-bit samples get shifted to 32 bits. */
|
|
|
+ if (format->encoding == PCM_CODE && format->bitspersample == 24) {
|
|
|
+ return PCM_ConvertSint24ToSint32(file, audio_buf, audio_len);
|
|
|
+ }
|
|
|
+
|
|
|
+ outputsize = (size_t)file->sampleframes;
|
|
|
+ if (SafeMult(&outputsize, format->blockalign)) {
|
|
|
+ return SDL_OutOfMemory();
|
|
|
+ } else if (outputsize > SDL_MAX_UINT32 || file->sampleframes > SIZE_MAX) {
|
|
|
+ return SDL_SetError("WAVE file too big");
|
|
|
+ }
|
|
|
+
|
|
|
+ *audio_buf = chunk->data;
|
|
|
+ *audio_len = (Uint32)outputsize;
|
|
|
+
|
|
|
+ /* This pointer is going to be returned to the caller. Prevent free in cleanup. */
|
|
|
+ chunk->data = NULL;
|
|
|
+ chunk->size = 0;
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static WaveRiffSizeHint
|
|
|
+WaveGetRiffSizeHint()
|
|
|
+{
|
|
|
+ const char *hint = SDL_GetHint(SDL_HINT_WAVE_RIFF_CHUNK_SIZE);
|
|
|
+
|
|
|
+ if (hint != NULL) {
|
|
|
+ if (SDL_strcmp(hint, "force") == 0) {
|
|
|
+ return RiffSizeForce;
|
|
|
+ } else if (SDL_strcmp(hint, "ignore") == 0) {
|
|
|
+ return RiffSizeIgnore;
|
|
|
+ } else if (SDL_strcmp(hint, "ignorezero") == 0) {
|
|
|
+ return RiffSizeIgnoreZero;
|
|
|
+ } else if (SDL_strcmp(hint, "maximum") == 0) {
|
|
|
+ return RiffSizeMaximum;
|
|
|
+ }
|
|
|
+ }
|
|
|
|
|
|
+ return RiffSizeNoHint;
|
|
|
+}
|
|
|
+
|
|
|
+static WaveTruncationHint
|
|
|
+WaveGetTruncationHint()
|
|
|
+{
|
|
|
+ const char *hint = SDL_GetHint(SDL_HINT_WAVE_TRUNCATION);
|
|
|
+
|
|
|
+ if (hint != NULL) {
|
|
|
+ if (SDL_strcmp(hint, "verystrict") == 0) {
|
|
|
+ return TruncVeryStrict;
|
|
|
+ } else if (SDL_strcmp(hint, "strict") == 0) {
|
|
|
+ return TruncStrict;
|
|
|
+ } else if (SDL_strcmp(hint, "dropframe") == 0) {
|
|
|
+ return TruncDropFrame;
|
|
|
+ } else if (SDL_strcmp(hint, "dropblock") == 0) {
|
|
|
+ return TruncDropBlock;
|
|
|
+ }
|
|
|
+ }
|
|
|
|
|
|
-static int ReadChunk(SDL_RWops * src, Chunk * chunk);
|
|
|
+ return TruncNoHint;
|
|
|
+}
|
|
|
|
|
|
-struct MS_ADPCM_decodestate
|
|
|
+static WaveFactChunkHint
|
|
|
+WaveGetFactChunkHint()
|
|
|
{
|
|
|
- Uint8 hPredictor;
|
|
|
- Uint16 iDelta;
|
|
|
- Sint16 iSamp1;
|
|
|
- Sint16 iSamp2;
|
|
|
-};
|
|
|
-static struct MS_ADPCM_decoder
|
|
|
+ const char *hint = SDL_GetHint(SDL_HINT_WAVE_FACT_CHUNK);
|
|
|
+
|
|
|
+ if (hint != NULL) {
|
|
|
+ if (SDL_strcmp(hint, "truncate") == 0) {
|
|
|
+ return FactTruncate;
|
|
|
+ } else if (SDL_strcmp(hint, "strict") == 0) {
|
|
|
+ return FactStrict;
|
|
|
+ } else if (SDL_strcmp(hint, "ignorezero") == 0) {
|
|
|
+ return FactIgnoreZero;
|
|
|
+ } else if (SDL_strcmp(hint, "ignore") == 0) {
|
|
|
+ return FactIgnore;
|
|
|
+ }
|
|
|
+ }
|
|
|
+
|
|
|
+ return FactNoHint;
|
|
|
+}
|
|
|
+
|
|
|
+static void
|
|
|
+WaveFreeChunkData(WaveChunk *chunk)
|
|
|
{
|
|
|
- WaveFMT wavefmt;
|
|
|
- Uint16 wSamplesPerBlock;
|
|
|
- Uint16 wNumCoef;
|
|
|
- Sint16 aCoeff[7][2];
|
|
|
- /* * * */
|
|
|
- struct MS_ADPCM_decodestate state[2];
|
|
|
-} MS_ADPCM_state;
|
|
|
+ if (chunk->data != NULL) {
|
|
|
+ SDL_free(chunk->data);
|
|
|
+ chunk->data = NULL;
|
|
|
+ }
|
|
|
+ chunk->size = 0;
|
|
|
+}
|
|
|
|
|
|
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[] = {
|
|
|
- 230, 230, 230, 230, 307, 409, 512, 614,
|
|
|
- 768, 614, 512, 409, 307, 230, 230, 230
|
|
|
- };
|
|
|
- Sint32 new_sample, delta;
|
|
|
+WaveNextChunk(SDL_RWops *src, WaveChunk *chunk)
|
|
|
+{
|
|
|
+ Uint32 chunkheader[2];
|
|
|
+ Sint64 nextposition = chunk->position + chunk->length;
|
|
|
|
|
|
- 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;
|
|
|
+ /* Data is no longer valid after this function returns. */
|
|
|
+ WaveFreeChunkData(chunk);
|
|
|
+
|
|
|
+ /* Error on overflows. */
|
|
|
+ if (SDL_MAX_SINT64 - chunk->length < chunk->position || SDL_MAX_SINT64 - 8 < nextposition) {
|
|
|
+ return -1;
|
|
|
}
|
|
|
- if (new_sample < min_audioval) {
|
|
|
- new_sample = min_audioval;
|
|
|
- } else if (new_sample > max_audioval) {
|
|
|
- new_sample = max_audioval;
|
|
|
+
|
|
|
+ /* RIFF chunks have a 2-byte alignment. Skip padding byte. */
|
|
|
+ if (chunk->length & 1) {
|
|
|
+ nextposition++;
|
|
|
}
|
|
|
- delta = ((Sint32) state->iDelta * adaptive[nybble]) / 256;
|
|
|
- if (delta < 16) {
|
|
|
- delta = 16;
|
|
|
+
|
|
|
+ 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;
|
|
|
}
|
|
|
- state->iDelta = (Uint16) delta;
|
|
|
- state->iSamp2 = state->iSamp1;
|
|
|
- state->iSamp1 = (Sint16) new_sample;
|
|
|
- return (new_sample);
|
|
|
+
|
|
|
+ chunk->fourcc = SDL_SwapLE32(chunkheader[0]);
|
|
|
+ chunk->length = SDL_SwapLE32(chunkheader[1]);
|
|
|
+ chunk->position = nextposition + 8;
|
|
|
+
|
|
|
+ return 0;
|
|
|
}
|
|
|
|
|
|
static int
|
|
|
-MS_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
|
|
|
-{
|
|
|
- struct MS_ADPCM_decodestate *state[2];
|
|
|
- Uint8 *freeable, *encoded, *decoded;
|
|
|
- Sint32 encoded_len, samplesleft;
|
|
|
- Sint8 nybble;
|
|
|
- Uint8 stereo;
|
|
|
- Sint16 *coeff[2];
|
|
|
- Sint32 new_sample;
|
|
|
+WaveReadPartialChunkData(SDL_RWops *src, WaveChunk *chunk, size_t length)
|
|
|
+{
|
|
|
+ WaveFreeChunkData(chunk);
|
|
|
|
|
|
- /* 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();
|
|
|
+ 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. */
|
|
|
+ }
|
|
|
}
|
|
|
- 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;
|
|
|
- }
|
|
|
- encoded_len -= MS_ADPCM_state.wavefmt.blockalign;
|
|
|
- }
|
|
|
- SDL_free(freeable);
|
|
|
- return (0);
|
|
|
-}
|
|
|
-
|
|
|
-struct IMA_ADPCM_decodestate
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+static int
|
|
|
+WaveReadChunkData(SDL_RWops *src, WaveChunk *chunk)
|
|
|
{
|
|
|
- Sint32 sample;
|
|
|
- Sint8 index;
|
|
|
+ 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 struct IMA_ADPCM_decoder
|
|
|
+
|
|
|
+static Uint16
|
|
|
+WaveGetFormatGUIDEncoding(WaveFormat *format)
|
|
|
{
|
|
|
- WaveFMT wavefmt;
|
|
|
- Uint16 wSamplesPerBlock;
|
|
|
- /* * * */
|
|
|
- struct IMA_ADPCM_decodestate state[2];
|
|
|
-} IMA_ADPCM_state;
|
|
|
+ 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
|
|
|
-InitIMA_ADPCM(WaveFMT * format)
|
|
|
+WaveReadFormat(WaveFile *file)
|
|
|
{
|
|
|
- Uint8 *rogue_feel;
|
|
|
+ WaveChunk *chunk = &file->chunk;
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
+ SDL_RWops *fmtsrc;
|
|
|
+ size_t fmtlen = chunk->size;
|
|
|
|
|
|
- /* Set the rogue pointer to the IMA_ADPCM specific data */
|
|
|
- IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
|
|
|
- IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
|
|
|
- IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
|
|
|
- IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
|
|
|
- IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
|
|
|
- IMA_ADPCM_state.wavefmt.bitspersample =
|
|
|
- SDL_SwapLE16(format->bitspersample);
|
|
|
- rogue_feel = (Uint8 *) format + sizeof(*format);
|
|
|
- if (sizeof(*format) == 16) {
|
|
|
- /* const Uint16 extra_info = ((rogue_feel[1] << 8) | rogue_feel[0]); */
|
|
|
- rogue_feel += sizeof(Uint16);
|
|
|
+ if (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();
|
|
|
}
|
|
|
- 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] = {
|
|
|
- -1, -1, -1, -1,
|
|
|
- 2, 4, 6, 8,
|
|
|
- -1, -1, -1, -1,
|
|
|
- 2, 4, 6, 8
|
|
|
- };
|
|
|
- const Sint32 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,
|
|
|
- 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,
|
|
|
- 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
|
|
|
- 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630,
|
|
|
- 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,
|
|
|
- 22385, 24623, 27086, 29794, 32767
|
|
|
- };
|
|
|
- Sint32 delta, step;
|
|
|
+ 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);
|
|
|
|
|
|
- /* Compute difference and new sample value */
|
|
|
- if (state->index > 88) {
|
|
|
- state->index = 88;
|
|
|
- } else if (state->index < 0) {
|
|
|
- state->index = 0;
|
|
|
+ /* 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");
|
|
|
}
|
|
|
- /* explicit cast to avoid gcc warning about using 'char' as array index */
|
|
|
- step = step_table[(int)state->index];
|
|
|
- delta = step >> 3;
|
|
|
- if (nybble & 0x04)
|
|
|
- delta += step;
|
|
|
- if (nybble & 0x02)
|
|
|
- delta += (step >> 1);
|
|
|
- if (nybble & 0x01)
|
|
|
- delta += (step >> 2);
|
|
|
- if (nybble & 0x08)
|
|
|
- delta = -delta;
|
|
|
- state->sample += delta;
|
|
|
|
|
|
- /* Update index value */
|
|
|
- state->index += index_table[nybble];
|
|
|
+ /* The earlier versions also don't have this field. */
|
|
|
+ if (fmtlen >= 18) {
|
|
|
+ format->extsize = SDL_ReadLE16(fmtsrc);
|
|
|
+ }
|
|
|
|
|
|
- /* Clamp output sample */
|
|
|
- if (state->sample > max_audioval) {
|
|
|
- state->sample = max_audioval;
|
|
|
- } else if (state->sample < min_audioval) {
|
|
|
- state->sample = min_audioval;
|
|
|
+ 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);
|
|
|
}
|
|
|
- return (state->sample);
|
|
|
+
|
|
|
+ SDL_RWclose(fmtsrc);
|
|
|
+
|
|
|
+ return 0;
|
|
|
}
|
|
|
|
|
|
-/* 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
|
|
|
+WaveCheckFormat(WaveFile *file, size_t datalength)
|
|
|
{
|
|
|
- int i;
|
|
|
- Sint8 nybble;
|
|
|
- Sint32 new_sample;
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
|
|
|
- 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;
|
|
|
+ /* Check for some obvious issues. */
|
|
|
|
|
|
- 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;
|
|
|
+ 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");
|
|
|
+ }
|
|
|
|
|
|
- ++encoded;
|
|
|
+ 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);
|
|
|
}
|
|
|
-}
|
|
|
|
|
|
-static int
|
|
|
-IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
|
|
|
-{
|
|
|
- struct IMA_ADPCM_decodestate *state;
|
|
|
- Uint8 *freeable, *encoded, *decoded;
|
|
|
- Sint32 encoded_len, samplesleft;
|
|
|
- unsigned int c, channels;
|
|
|
-
|
|
|
- /* Check to make sure we have enough variables in the state array */
|
|
|
- channels = IMA_ADPCM_state.wavefmt.channels;
|
|
|
- if (channels > SDL_arraysize(IMA_ADPCM_state.state)) {
|
|
|
- SDL_SetError("IMA ADPCM decoder can only handle %u channels",
|
|
|
- (unsigned int)SDL_arraysize(IMA_ADPCM_state.state));
|
|
|
- return (-1);
|
|
|
- }
|
|
|
- state = IMA_ADPCM_state.state;
|
|
|
-
|
|
|
- /* Allocate the proper sized output buffer */
|
|
|
- encoded_len = *audio_len;
|
|
|
- encoded = *audio_buf;
|
|
|
- freeable = *audio_buf;
|
|
|
- *audio_len = (encoded_len / IMA_ADPCM_state.wavefmt.blockalign) *
|
|
|
- IMA_ADPCM_state.wSamplesPerBlock *
|
|
|
- IMA_ADPCM_state.wavefmt.channels * sizeof(Sint16);
|
|
|
- *audio_buf = (Uint8 *) SDL_malloc(*audio_len);
|
|
|
- if (*audio_buf == NULL) {
|
|
|
- return SDL_OutOfMemory();
|
|
|
+ /* Reject invalid fact chunks in strict mode. */
|
|
|
+ if (file->facthint == FactStrict && file->fact.status == -1) {
|
|
|
+ return SDL_SetError("Invalid fact chunk in WAVE file");
|
|
|
}
|
|
|
- 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? */ ;
|
|
|
- }
|
|
|
|
|
|
- /* Store the initial sample we start with */
|
|
|
- decoded[0] = (Uint8) (state[c].sample & 0xFF);
|
|
|
- decoded[1] = (Uint8) (state[c].sample >> 8);
|
|
|
- decoded += 2;
|
|
|
+ /* Check for issues common to all encodings. Some unsupported formats set
|
|
|
+ * the bits per sample to zero. These fall through to the 'unsupported
|
|
|
+ * format' error.
|
|
|
+ */
|
|
|
+ switch (format->encoding) {
|
|
|
+ case IEEE_FLOAT_CODE:
|
|
|
+ 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");
|
|
|
}
|
|
|
|
|
|
- /* 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;
|
|
|
+ /* 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;
|
|
|
}
|
|
|
- decoded += (channels * 8 * 2);
|
|
|
}
|
|
|
- encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
|
|
|
}
|
|
|
- SDL_free(freeable);
|
|
|
- return (0);
|
|
|
-}
|
|
|
|
|
|
+ /* Check the format for encoding specific issues and initialize decoders. */
|
|
|
+ switch (format->encoding) {
|
|
|
+ case PCM_CODE:
|
|
|
+ case IEEE_FLOAT_CODE:
|
|
|
+ if (PCM_Init(file, datalength) < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+ break;
|
|
|
+ case ALAW_CODE:
|
|
|
+ case MULAW_CODE:
|
|
|
+ if (LAW_Init(file, datalength) < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+ break;
|
|
|
+ case MS_ADPCM_CODE:
|
|
|
+ if (MS_ADPCM_Init(file, datalength) < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
+ break;
|
|
|
+ case IMA_ADPCM_CODE:
|
|
|
+ if (IMA_ADPCM_Init(file, datalength) < 0) {
|
|
|
+ return -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", (unsigned int)format->encoding);
|
|
|
+ }
|
|
|
+
|
|
|
+ return 0;
|
|
|
+}
|
|
|
|
|
|
static int
|
|
|
-ConvertSint24ToSint32(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;
|
|
|
-
|
|
|
- if (!ptr) {
|
|
|
- return SDL_OutOfMemory();
|
|
|
+WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
|
|
|
+{
|
|
|
+ int result;
|
|
|
+ Uint32 chunkcount = 0;
|
|
|
+ Uint32 chunkcountlimit = 10000;
|
|
|
+ char *envchunkcountlimit;
|
|
|
+ Sint64 RIFFstart, RIFFend, lastchunkpos;
|
|
|
+ SDL_bool RIFFlengthknown = SDL_FALSE;
|
|
|
+ WaveFormat *format = &file->format;
|
|
|
+ WaveChunk *chunk = &file->chunk;
|
|
|
+ WaveChunk RIFFchunk;
|
|
|
+ WaveChunk fmtchunk;
|
|
|
+ WaveChunk datachunk;
|
|
|
+
|
|
|
+ SDL_zero(RIFFchunk);
|
|
|
+ SDL_zero(fmtchunk);
|
|
|
+ SDL_zero(datachunk);
|
|
|
+
|
|
|
+ envchunkcountlimit = SDL_getenv("SDL_WAVE_CHUNK_LIMIT");
|
|
|
+ if (envchunkcountlimit != NULL) {
|
|
|
+ unsigned int count;
|
|
|
+ if (SDL_sscanf(envchunkcountlimit, "%u", &count) == 1) {
|
|
|
+ chunkcountlimit = count <= SDL_MAX_UINT32 ? count : SDL_MAX_UINT32;
|
|
|
+ }
|
|
|
}
|
|
|
|
|
|
- *audio_buf = ptr;
|
|
|
- *audio_len = expanded_len;
|
|
|
+ RIFFstart = SDL_RWtell(src);
|
|
|
+ if (RIFFstart < 0) {
|
|
|
+ return SDL_SetError("Could not seek in file");
|
|
|
+ }
|
|
|
|
|
|
- /* 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);
|
|
|
+ RIFFchunk.position = RIFFstart;
|
|
|
+ if (WaveNextChunk(src, &RIFFchunk) < 0) {
|
|
|
+ return SDL_SetError("Could not read RIFF header");
|
|
|
}
|
|
|
|
|
|
- return 0;
|
|
|
-}
|
|
|
+ /* 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;
|
|
|
|
|
|
-/* 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 };
|
|
|
+ /* Use the RIFF chunk size to limit the search for the chunks. This is not
|
|
|
+ * always reliable and the hint can be used to tune the behavior. By
|
|
|
+ * default, it will never search past 4 GiB.
|
|
|
+ */
|
|
|
+ switch (file->riffhint) {
|
|
|
+ case RiffSizeIgnore:
|
|
|
+ RIFFend = RIFFchunk.position + SDL_MAX_UINT32;
|
|
|
+ break;
|
|
|
+ default:
|
|
|
+ case RiffSizeIgnoreZero:
|
|
|
+ if (RIFFchunk.length == 0) {
|
|
|
+ RIFFend = RIFFchunk.position + SDL_MAX_UINT32;
|
|
|
+ break;
|
|
|
+ }
|
|
|
+ /* fallthrough */
|
|
|
+ case RiffSizeForce:
|
|
|
+ RIFFend = RIFFchunk.position + RIFFchunk.length;
|
|
|
+ RIFFlengthknown = SDL_TRUE;
|
|
|
+ break;
|
|
|
+ case RiffSizeMaximum:
|
|
|
+ RIFFend = SDL_MAX_SINT64;
|
|
|
+ break;
|
|
|
+ }
|
|
|
|
|
|
-SDL_AudioSpec *
|
|
|
-SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
|
|
|
- SDL_AudioSpec * spec, Uint8 ** audio_buf, Uint32 * audio_len)
|
|
|
-{
|
|
|
- int was_error;
|
|
|
- Chunk chunk;
|
|
|
- int lenread;
|
|
|
- int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
|
|
|
- int samplesize;
|
|
|
+ /* Step through all chunks and save information on the fmt, data, and fact
|
|
|
+ * chunks. Ignore the chunks we don't know as per specification. This
|
|
|
+ * currently also ignores cue, list, and slnt chunks.
|
|
|
+ */
|
|
|
+ while ((Uint64)RIFFend > (Uint64)chunk->position + chunk->length + (chunk->length & 1)) {
|
|
|
+ /* Abort after too many chunks or else corrupt files may waste time. */
|
|
|
+ if (chunkcount++ >= chunkcountlimit) {
|
|
|
+ return SDL_SetError("Chunk count in WAVE file exceeds limit of %u", chunkcountlimit);
|
|
|
+ }
|
|
|
+
|
|
|
+ result = WaveNextChunk(src, chunk);
|
|
|
+ if (result == -1) {
|
|
|
+ /* Unexpected EOF. Corrupt file or I/O issues. */
|
|
|
+ if (file->trunchint == TruncVeryStrict) {
|
|
|
+ return SDL_SetError("Unexpected end of WAVE file");
|
|
|
+ }
|
|
|
+ /* Let the checks after this loop sort this issue out. */
|
|
|
+ break;
|
|
|
+ } else if (result == -2) {
|
|
|
+ return SDL_SetError("Could not seek to WAVE chunk header");
|
|
|
+ }
|
|
|
|
|
|
- /* WAV magic header */
|
|
|
- Uint32 RIFFchunk;
|
|
|
- Uint32 wavelen = 0;
|
|
|
- Uint32 WAVEmagic;
|
|
|
- Uint32 headerDiff = 0;
|
|
|
+ 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;
|
|
|
+ }
|
|
|
+ }
|
|
|
+ }
|
|
|
+ }
|
|
|
|
|
|
- /* FMT chunk */
|
|
|
- WaveFMT *format = NULL;
|
|
|
- WaveExtensibleFMT *ext = NULL;
|
|
|
+ /* Go through all chunks in verystrict mode or stop the search early if
|
|
|
+ * all required chunks were found.
|
|
|
+ */
|
|
|
+ if (file->trunchint == TruncVeryStrict) {
|
|
|
+ if ((Uint64)RIFFend < (Uint64)chunk->position + chunk->length) {
|
|
|
+ return SDL_SetError("RIFF size truncates chunk");
|
|
|
+ }
|
|
|
+ } else if (fmtchunk.fourcc == FMT && datachunk.fourcc == DATA) {
|
|
|
+ if (file->fact.status == 1 || file->facthint == FactIgnore || file->facthint == FactNoHint) {
|
|
|
+ break;
|
|
|
+ }
|
|
|
+ }
|
|
|
+ }
|
|
|
|
|
|
- SDL_zero(chunk);
|
|
|
+ /* Save the position after the last chunk. This position will be used if the
|
|
|
+ * RIFF length is unknown.
|
|
|
+ */
|
|
|
+ lastchunkpos = chunk->position + chunk->length;
|
|
|
|
|
|
- /* Make sure we are passed a valid data source */
|
|
|
- was_error = 0;
|
|
|
- if (src == NULL) {
|
|
|
- was_error = 1;
|
|
|
- goto done;
|
|
|
+ /* The fmt chunk is mandatory. */
|
|
|
+ if (fmtchunk.fourcc != FMT) {
|
|
|
+ return SDL_SetError("Missing fmt chunk in WAVE file");
|
|
|
+ }
|
|
|
+ /* A data chunk must be present. */
|
|
|
+ if (datachunk.fourcc != DATA) {
|
|
|
+ return SDL_SetError("Missing data chunk in WAVE file");
|
|
|
+ }
|
|
|
+ /* Check if the last chunk has all of its data in verystrict mode. */
|
|
|
+ if (file->trunchint == TruncVeryStrict) {
|
|
|
+ /* data chunk is handled later. */
|
|
|
+ if (chunk->fourcc != DATA && chunk->length > 0) {
|
|
|
+ Uint8 tmp;
|
|
|
+ Uint64 position = (Uint64)chunk->position + chunk->length - 1;
|
|
|
+ if (position > SDL_MAX_SINT64 || SDL_RWseek(src, (Sint64)position, RW_SEEK_SET) != (Sint64)position) {
|
|
|
+ return SDL_SetError("Could not seek to WAVE chunk data");
|
|
|
+ } else if (SDL_RWread(src, &tmp, 1, 1) != 1) {
|
|
|
+ return SDL_SetError("RIFF size truncates chunk");
|
|
|
+ }
|
|
|
+ }
|
|
|
}
|
|
|
|
|
|
- /* Check the magic header */
|
|
|
- RIFFchunk = SDL_ReadLE32(src);
|
|
|
- wavelen = SDL_ReadLE32(src);
|
|
|
- if (wavelen == WAVE) { /* The RIFFchunk has already been read */
|
|
|
- WAVEmagic = wavelen;
|
|
|
- wavelen = RIFFchunk;
|
|
|
- RIFFchunk = RIFF;
|
|
|
- } else {
|
|
|
- WAVEmagic = SDL_ReadLE32(src);
|
|
|
- }
|
|
|
- if ((RIFFchunk != RIFF) || (WAVEmagic != WAVE)) {
|
|
|
- SDL_SetError("Unrecognized file type (not WAVE)");
|
|
|
- was_error = 1;
|
|
|
- goto done;
|
|
|
- }
|
|
|
- headerDiff += sizeof(Uint32); /* for WAVE */
|
|
|
-
|
|
|
- /* 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;
|
|
|
- }
|
|
|
- IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
|
|
|
- switch (SDL_SwapLE16(format->encoding)) {
|
|
|
+ /* 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:
|
|
|
- /* We can understand this */
|
|
|
- break;
|
|
|
case IEEE_FLOAT_CODE:
|
|
|
- IEEE_float_encoded = 1;
|
|
|
- /* We can understand this */
|
|
|
+ if (PCM_Decode(file, audio_buf, audio_len) < 0) {
|
|
|
+ return -1;
|
|
|
+ }
|
|
|
break;
|
|
|
- case MS_ADPCM_CODE:
|
|
|
- /* Try to understand this */
|
|
|
- if (InitMS_ADPCM(format) < 0) {
|
|
|
- was_error = 1;
|
|
|
- goto done;
|
|
|
+ case ALAW_CODE:
|
|
|
+ case MULAW_CODE:
|
|
|
+ if (LAW_Decode(file, audio_buf, audio_len) < 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;
|
|
|
+ case MS_ADPCM_CODE:
|
|
|
+ if (MS_ADPCM_Decode(file, audio_buf, audio_len) < 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 IMA_ADPCM_CODE:
|
|
|
+ if (IMA_ADPCM_Decode(file, audio_buf, audio_len) < 0) {
|
|
|
+ return -1;
|
|
|
}
|
|
|
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;
|
|
|
}
|
|
|
+
|
|
|
+ /* Setting up the SDL_AudioSpec. All unsupported formats were filtered out
|
|
|
+ * by checks earlier in this function.
|
|
|
+ */
|
|
|
SDL_zerop(spec);
|
|
|
- spec->freq = SDL_SwapLE32(format->frequency);
|
|
|
+ spec->freq = format->frequency;
|
|
|
+ spec->channels = (Uint8)format->channels;
|
|
|
+ spec->samples = 4096; /* Good default buffer size */
|
|
|
|
|
|
- if (IEEE_float_encoded) {
|
|
|
- if ((SDL_SwapLE16(format->bitspersample)) != 32) {
|
|
|
- was_error = 1;
|
|
|
- } else {
|
|
|
- spec->format = AUDIO_F32;
|
|
|
- }
|
|
|
- } else {
|
|
|
- switch (SDL_SwapLE16(format->bitspersample)) {
|
|
|
- case 4:
|
|
|
- if (MS_ADPCM_encoded || IMA_ADPCM_encoded) {
|
|
|
- spec->format = AUDIO_S16;
|
|
|
- } else {
|
|
|
- was_error = 1;
|
|
|
- }
|
|
|
- break;
|
|
|
+ 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 %u-bit PCM data format", (unsigned int)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;
|
|
|
|
|
|
- if (SDL_SwapLE16(format->bitspersample) == 24) {
|
|
|
- if (ConvertSint24ToSint32(audio_buf, audio_len) < 0) {
|
|
|
- was_error = 1;
|
|
|
- goto done;
|
|
|
- }
|
|
|
+ SDL_zero(file);
|
|
|
+
|
|
|
+ /* Make sure we are passed a valid data source */
|
|
|
+ if (src == NULL) {
|
|
|
+ /* Error may come from RWops. */
|
|
|
+ return NULL;
|
|
|
+ } else if (spec == NULL) {
|
|
|
+ SDL_InvalidParamError("spec");
|
|
|
+ return NULL;
|
|
|
+ } else if (audio_buf == NULL) {
|
|
|
+ SDL_InvalidParamError("audio_buf");
|
|
|
+ return NULL;
|
|
|
+ } else if (audio_len == NULL) {
|
|
|
+ SDL_InvalidParamError("audio_len");
|
|
|
+ return NULL;
|
|
|
}
|
|
|
|
|
|
- /* 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;
|
|
|
|
|
|
- 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) {
|
|
|
+ file.riffhint = WaveGetRiffSizeHint();
|
|
|
+ file.trunchint = WaveGetTruncationHint();
|
|
|
+ file.facthint = WaveGetFactChunkHint();
|
|
|
+
|
|
|
+ result = WaveLoad(src, &file, spec, audio_buf, audio_len);
|
|
|
+ if (result < 0) {
|
|
|
+ SDL_free(*audio_buf);
|
|
|
spec = NULL;
|
|
|
+ audio_buf = NULL;
|
|
|
+ audio_len = 0;
|
|
|
+ }
|
|
|
+
|
|
|
+ /* Cleanup */
|
|
|
+ if (freesrc) {
|
|
|
+ SDL_RWclose(src);
|
|
|
+ } else {
|
|
|
+ SDL_RWseek(src, file.chunk.position, RW_SEEK_SET);
|
|
|
}
|
|
|
- return (spec);
|
|
|
+ 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: */
|
woollybah