# HG changeset patch # User Ryan C. Gordon # Date 1009494305 0 # Node ID 6bb68b3bdcf1ff5c1b2df356aa632a56b25412c4 # Parent 47cc2de2ae36031ab800615a4a6f78c642a8d753 ADPCM decoder seems to be complete, other fixes too... diff -r 47cc2de2ae36 -r 6bb68b3bdcf1 decoders/wav.c --- a/decoders/wav.c Wed Dec 26 10:40:25 2001 +0000 +++ b/decoders/wav.c Thu Dec 27 23:05:05 2001 +0000 @@ -101,6 +101,7 @@ #define riffID 0x46464952 /* "RIFF", in ascii. */ #define waveID 0x45564157 /* "WAVE", in ascii. */ +#define factID 0x74636166 /* "fact", in ascii. */ /***************************************************************************** @@ -114,8 +115,8 @@ typedef struct { - Uint16 iCoef1; - Uint16 iCoef2; + Sint16 iCoef1; + Sint16 iCoef2; } ADPCMCOEFSET; typedef struct @@ -137,6 +138,8 @@ Uint16 wBlockAlign; Uint16 wBitsPerSample; + Uint32 sample_frame_size; + void (*free)(struct S_WAV_FMT_T *fmt); Uint32 (*read_sample)(Sound_Sample *sample); @@ -149,6 +152,9 @@ Uint16 wNumCoef; ADPCMCOEFSET *aCoef; ADPCMBLOCKHEADER *blockheaders; + Uint32 samples_left_in_block; + int nibble_state; + Sint8 nibble; } adpcm; /* put other format-specific data here... */ @@ -226,6 +232,9 @@ * Normal, uncompressed waveform handler... * *****************************************************************************/ +/* + * Sound_Decode() lands here for uncompressed WAVs... + */ static Uint32 read_sample_fmt_normal(Sound_Sample *sample) { Uint32 retval; @@ -274,12 +283,29 @@ * ADPCM compression handler... * *****************************************************************************/ -static int read_adpcm_block_headers(SDL_RWops *rw, fmt_t *fmt) +#define FIXED_POINT_COEF_BASE 256 +#define FIXED_POINT_ADAPTION_BASE 256 +#define SMALLEST_ADPCM_DELTA 16 + + +static inline int read_adpcm_block_headers(Sound_Sample *sample) { + Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; + SDL_RWops *rw = internal->rw; + wav_t *w = (wav_t *) internal->decoder_private; + fmt_t *fmt = w->fmt; ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; int i; int max = fmt->wChannels; + if (w->bytesLeft < fmt->wBlockAlign) + { + sample->flags |= SOUND_SAMPLEFLAG_EOF; + return(0); + } /* if */ + + w->bytesLeft -= fmt->wBlockAlign; + for (i = 0; i < max; i++) BAIL_IF_MACRO(!read_uint8(rw, &headers[i].bPredictor), NULL, 0); @@ -292,48 +318,164 @@ for (i = 0; i < max; i++) BAIL_IF_MACRO(!read_le16(rw, &headers[i].iSamp2), NULL, 0); + fmt->fmt.adpcm.samples_left_in_block = fmt->fmt.adpcm.wSamplesPerBlock; + fmt->fmt.adpcm.nibble_state = 0; return(1); } /* read_adpcm_block_headers */ -static int decode_adpcm_block(Sound_Sample *sample) +static inline void do_adpcm_nibble(Uint8 nib, + ADPCMBLOCKHEADER *header, + Sint32 lPredSamp) +{ + static const Sint32 max_audioval = ((1<<(16-1))-1); + static const Sint32 min_audioval = -(1<<(16-1)); + static const Sint32 AdaptionTable[] = + { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 + }; + + Sint32 lNewSamp; + Sint32 delta; + + if (nib & 0x08) + lNewSamp = lPredSamp + (header->iDelta * (nib - 0x10)); + else + lNewSamp = lPredSamp + (header->iDelta * nib); + + /* clamp value... */ + if (lNewSamp < min_audioval) + lNewSamp = min_audioval; + else if (lNewSamp > max_audioval) + lNewSamp = max_audioval; + + delta = ((Sint32) header->iDelta * AdaptionTable[nib]) / + FIXED_POINT_ADAPTION_BASE; + + if (delta < SMALLEST_ADPCM_DELTA) + delta = SMALLEST_ADPCM_DELTA; + + header->iDelta = delta; + header->iSamp2 = header->iSamp1; + header->iSamp1 = lNewSamp; +} /* do_adpcm_nibble */ + + +static inline int decode_adpcm_sample_frame(Sound_Sample *sample) { Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; - SDL_RWops *rw = internal->rw; wav_t *w = (wav_t *) internal->decoder_private; fmt_t *fmt = w->fmt; ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; + SDL_RWops *rw = internal->rw; int i; int max = fmt->wChannels; - - if (!read_adpcm_block_headers(rw, fmt)) - { - sample->flags |= SOUND_SAMPLEFLAG_ERROR; - return(0); - } /* if */ + Sint32 delta; + Uint8 nib = fmt->fmt.adpcm.nibble; for (i = 0; i < max; i++) { - Uint16 iCoef1 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef1; - Uint16 iCoef2 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef2; -/* + Uint8 byte; + Sint16 iCoef1 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef1; + Sint16 iCoef2 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef2; Sint32 lPredSamp = ((headers[i].iSamp1 * iCoef1) + (headers[i].iSamp2 * iCoef2)) / - FIXED_POINT_COEF_BASE; -*/ + FIXED_POINT_COEF_BASE; + + if (fmt->fmt.adpcm.nibble_state == 0) + { + BAIL_IF_MACRO(!read_uint8(rw, &nib), NULL, 0); + fmt->fmt.adpcm.nibble_state = 1; + do_adpcm_nibble(nib >> 4, &headers[i], lPredSamp); + } /* if */ + else + { + fmt->fmt.adpcm.nibble_state = 0; + do_adpcm_nibble(nib & 0x0F, &headers[i], lPredSamp); + } /* else */ } /* for */ -} /* decode_adpcm_block */ + + fmt->fmt.adpcm.nibble = nib; + return(1); +} /* decode_adpcm_sample_frame */ + + +static inline void put_adpcm_sample_frame1(Uint8 *_buf, fmt_t *fmt) +{ + Uint16 *buf = (Uint16 *) _buf; + ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; + int i; + for (i = 0; i < fmt->wChannels; i++) + *(buf++) = headers[i].iSamp1; +} /* put_adpcm_sample_frame1 */ + + +static inline void put_adpcm_sample_frame2(Uint8 *_buf, fmt_t *fmt) +{ + Uint16 *buf = (Uint16 *) _buf; + ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; + int i; + for (i = 0; i < fmt->wChannels; i++) + *(buf++) = headers[i].iSamp2; +} /* put_adpcm_sample_frame2 */ +/* + * Sound_Decode() lands here for ADPCM-encoded WAVs... + */ static Uint32 read_sample_fmt_adpcm(Sound_Sample *sample) { - /* !!! FIXME: Write this. */ - Sound_SetError("WAV: Not implemented. Soon."); - sample->flags |= SOUND_SAMPLEFLAG_ERROR; - return(0); + Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; + wav_t *w = (wav_t *) internal->decoder_private; + fmt_t *fmt = w->fmt; + Uint32 bw = 0; + + while (bw < internal->buffer_size) + { + /* write ongoing sample frame before reading more data... */ + switch (fmt->fmt.adpcm.samples_left_in_block) + { + case 0: /* need to read a new block... */ + if (!read_adpcm_block_headers(sample)) + { + if ((sample->flags & SOUND_SAMPLEFLAG_EOF) == 0) + sample->flags |= SOUND_SAMPLEFLAG_ERROR; + return(bw); + } /* if */ + + /* only write first sample frame for now. */ + put_adpcm_sample_frame2(internal->buffer + bw, fmt); + fmt->fmt.adpcm.samples_left_in_block--; + bw += fmt->sample_frame_size; + break; + + case 1: /* output last sample frame of block... */ + put_adpcm_sample_frame1(internal->buffer + bw, fmt); + fmt->fmt.adpcm.samples_left_in_block--; + bw += fmt->sample_frame_size; + break; + + default: /* output latest sample frame and read a new one... */ + put_adpcm_sample_frame1(internal->buffer + bw, fmt); + fmt->fmt.adpcm.samples_left_in_block--; + bw += fmt->sample_frame_size; + + if (!decode_adpcm_sample_frame(sample)) + { + sample->flags |= SOUND_SAMPLEFLAG_ERROR; + return(bw); + } /* if */ + } /* switch */ + } /* while */ + + return(bw); } /* read_sample_fmt_adpcm */ +/* + * Sound_FreeSample() lands here for ADPCM-encoded WAVs... + */ static void free_fmt_adpcm(fmt_t *fmt) { if (fmt->fmt.adpcm.aCoef != NULL) @@ -361,37 +503,22 @@ BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wSamplesPerBlock), NULL, 0); BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wNumCoef), NULL, 0); - /* - * !!! FIXME: SDL seems nervous about this, so I'll make a debug - * !!! FIXME: note for the time being... - */ - if (fmt->fmt.adpcm.wNumCoef != 7) - { - SNDDBG(("WAV: adpcm's wNumCoef is NOT seven (it's %d)!\n", - fmt->fmt.adpcm.wNumCoef)); - } /* if */ + /* fmt->free() is always called, so these malloc()s will be cleaned up. */ - /* fmt->free() is always called, so these malloc()s will be cleaned up. */ i = sizeof (ADPCMCOEFSET) * fmt->fmt.adpcm.wNumCoef; fmt->fmt.adpcm.aCoef = (ADPCMCOEFSET *) malloc(i); BAIL_IF_MACRO(fmt->fmt.adpcm.aCoef == NULL, ERR_OUT_OF_MEMORY, 0); + for (i = 0; i < fmt->fmt.adpcm.wNumCoef; i++) + { + BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef1), NULL, 0); + BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef2), NULL, 0); + } /* for */ + i = sizeof (ADPCMBLOCKHEADER) * fmt->wChannels; fmt->fmt.adpcm.blockheaders = (ADPCMBLOCKHEADER *) malloc(i); BAIL_IF_MACRO(fmt->fmt.adpcm.blockheaders == NULL, ERR_OUT_OF_MEMORY, 0); - for (i = 0; i < fmt->fmt.adpcm.wNumCoef; i++) - { - int rc; - rc = SDL_RWread(rw, &fmt->fmt.adpcm.aCoef[i].iCoef1, - sizeof (fmt->fmt.adpcm.aCoef[i].iCoef1), 1); - BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0); - - rc = SDL_RWread(rw, &fmt->fmt.adpcm.aCoef[i].iCoef2, - sizeof (fmt->fmt.adpcm.aCoef[i].iCoef2), 1); - BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0); - } /* for */ - return(1); } /* read_fmt_adpcm */ @@ -447,6 +574,7 @@ { Sint32 siz = 0; Uint32 _id = 0; + Uint32 pos = SDL_RWtell(rw); while (1) { @@ -456,8 +584,10 @@ /* skip ahead and see what next chunk is... */ BAIL_IF_MACRO(!read_le32(rw, &siz), NULL, 0); - assert(siz > 0); - BAIL_IF_MACRO(SDL_RWseek(rw, siz, SEEK_SET) != siz, NULL, 0); + assert(siz >= 0); + pos += (sizeof (Uint32) * 2) + siz; + if (siz > 0) + BAIL_IF_MACRO(SDL_RWseek(rw, pos, SEEK_SET) != pos, NULL, 0); } /* while */ return(0); /* shouldn't hit this, but just in case... */ @@ -470,6 +600,7 @@ SDL_RWops *rw = internal->rw; data_t d; wav_t *w; + Uint32 pos; BAIL_IF_MACRO(SDL_ReadLE32(rw) != riffID, "WAV: Not a RIFF file.", 0); SDL_ReadLE32(rw); /* throw the length away; we get this info later. */ @@ -479,9 +610,11 @@ sample->actual.channels = (Uint8) fmt->wChannels; sample->actual.rate = fmt->dwSamplesPerSec; - if (fmt->wBitsPerSample <= 8) + if (fmt->wBitsPerSample == 4) + sample->actual.format = AUDIO_S16SYS; /* !!! FIXME ? */ + else if (fmt->wBitsPerSample == 8) sample->actual.format = AUDIO_U8; - else if (fmt->wBitsPerSample <= 16) + else if (fmt->wBitsPerSample == 16) sample->actual.format = AUDIO_S16LSB; else BAIL_MACRO("WAV: Unsupported sample size.", 0); @@ -494,6 +627,11 @@ BAIL_IF_MACRO(w == NULL, ERR_OUT_OF_MEMORY, 0); w->fmt = fmt; w->bytesLeft = d.chunkSize; + + /* !!! FIXME: Move this to Sound_SampleInfo ? */ + fmt->sample_frame_size = ( ((sample->actual.format & 0xFF) / 8) * + sample->actual.channels ); + internal->decoder_private = (void *) w; sample->flags = SOUND_SAMPLEFLAG_NONE;