aboutsummaryrefslogtreecommitdiffstats
path: root/distrib/sdl-1.2.12/src/audio/SDL_wave.c
diff options
context:
space:
mode:
Diffstat (limited to 'distrib/sdl-1.2.12/src/audio/SDL_wave.c')
-rw-r--r--distrib/sdl-1.2.12/src/audio/SDL_wave.c597
1 files changed, 0 insertions, 597 deletions
diff --git a/distrib/sdl-1.2.12/src/audio/SDL_wave.c b/distrib/sdl-1.2.12/src/audio/SDL_wave.c
deleted file mode 100644
index 465195e..0000000
--- a/distrib/sdl-1.2.12/src/audio/SDL_wave.c
+++ /dev/null
@@ -1,597 +0,0 @@
-/*
- SDL - Simple DirectMedia Layer
- Copyright (C) 1997-2006 Sam Lantinga
-
- This library is free software; you can redistribute it and/or
- modify it under the terms of the GNU Lesser General Public
- License as published by the Free Software Foundation; either
- version 2.1 of the License, or (at your option) any later version.
-
- This library is distributed in the hope that it will be useful,
- but WITHOUT ANY WARRANTY; without even the implied warranty of
- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- Lesser General Public License for more details.
-
- You should have received a copy of the GNU Lesser General Public
- License along with this library; if not, write to the Free Software
- Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
-
- Sam Lantinga
- slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Microsoft WAVE file loading routines */
-
-#include "SDL_audio.h"
-#include "SDL_wave.h"
-
-
-static int ReadChunk(SDL_RWops *src, Chunk *chunk);
-
-struct MS_ADPCM_decodestate {
- Uint8 hPredictor;
- Uint16 iDelta;
- Sint16 iSamp1;
- Sint16 iSamp2;
-};
-static struct MS_ADPCM_decoder {
- WaveFMT wavefmt;
- Uint16 wSamplesPerBlock;
- Uint16 wNumCoef;
- Sint16 aCoeff[7][2];
- /* * * */
- struct MS_ADPCM_decodestate state[2];
-} MS_ADPCM_state;
-
-static int InitMS_ADPCM(WaveFMT *format)
-{
- Uint8 *rogue_feel;
- Uint16 extra_info;
- 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 ) {
- 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;
-
- 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;
- }
- if ( new_sample < min_audioval ) {
- new_sample = min_audioval;
- } else
- if ( new_sample > max_audioval ) {
- new_sample = max_audioval;
- }
- delta = ((Sint32)state->iDelta * adaptive[nybble])/256;
- if ( delta < 16 ) {
- delta = 16;
- }
- state->iDelta = (Uint16)delta;
- state->iSamp2 = state->iSamp1;
- state->iSamp1 = (Sint16)new_sample;
- return(new_sample);
-}
-
-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, stereo;
- Sint16 *coeff[2];
- Sint32 new_sample;
-
- /* 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 ) {
- SDL_Error(SDL_ENOMEM);
- return(-1);
- }
- 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 {
- Sint32 sample;
- Sint8 index;
-};
-static struct IMA_ADPCM_decoder {
- WaveFMT wavefmt;
- Uint16 wSamplesPerBlock;
- /* * * */
- struct IMA_ADPCM_decodestate state[2];
-} IMA_ADPCM_state;
-
-static int InitIMA_ADPCM(WaveFMT *format)
-{
- Uint8 *rogue_feel;
- Uint16 extra_info;
-
- /* 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 ) {
- extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);
- rogue_feel += sizeof(Uint16);
- }
- IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);
- return(0);
-}
-
-static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble)
-{
- const Sint32 max_audioval = ((1<<(16-1))-1);
- const Sint32 min_audioval = -(1<<(16-1));
- const int index_table[16] = {
- -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;
-
- /* Compute difference and new sample value */
- step = step_table[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];
- if ( state->index > 88 ) {
- state->index = 88;
- } else
- if ( state->index < 0 ) {
- state->index = 0;
- }
-
- /* Clamp output sample */
- if ( state->sample > max_audioval ) {
- state->sample = max_audioval;
- } else
- if ( state->sample < min_audioval ) {
- state->sample = min_audioval;
- }
- return(state->sample);
-}
-
-/* Fill the decode buffer with a channel block of data (8 samples) */
-static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded,
- int channel, int numchannels, struct IMA_ADPCM_decodestate *state)
-{
- int i;
- Sint8 nybble;
- Sint32 new_sample;
-
- decoded += (channel * 2);
- for ( i=0; i<4; ++i ) {
- nybble = (*encoded)&0x0F;
- new_sample = IMA_ADPCM_nibble(state, nybble);
- decoded[0] = new_sample&0xFF;
- new_sample >>= 8;
- decoded[1] = new_sample&0xFF;
- decoded += 2 * numchannels;
-
- nybble = (*encoded)>>4;
- new_sample = IMA_ADPCM_nibble(state, nybble);
- decoded[0] = new_sample&0xFF;
- new_sample >>= 8;
- decoded[1] = new_sample&0xFF;
- decoded += 2 * numchannels;
-
- ++encoded;
- }
-}
-
-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 %d channels",
- 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 ) {
- SDL_Error(SDL_ENOMEM);
- return(-1);
- }
- 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;
- }
-
- /* Decode and store the other samples in this block */
- samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels;
- while ( samplesleft > 0 ) {
- for ( c=0; c<channels; ++c ) {
- Fill_IMA_ADPCM_block(decoded, encoded,
- c, channels, &state[c]);
- encoded += 4;
- samplesleft -= 8;
- }
- decoded += (channels * 8 * 2);
- }
- encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
- }
- SDL_free(freeable);
- return(0);
-}
-
-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 MS_ADPCM_encoded, IMA_ADPCM_encoded;
- int samplesize;
-
- /* WAV magic header */
- Uint32 RIFFchunk;
- Uint32 wavelen = 0;
- Uint32 WAVEmagic;
- Uint32 headerDiff = 0;
-
- /* FMT chunk */
- WaveFMT *format = NULL;
-
- /* Make sure we are passed a valid data source */
- was_error = 0;
- if ( src == NULL ) {
- was_error = 1;
- goto done;
- }
-
- /* 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 {
- if ( chunk.data != NULL ) {
- SDL_free(chunk.data);
- }
- 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) );
-
- /* 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;
- }
- MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
- switch (SDL_SwapLE16(format->encoding)) {
- case PCM_CODE:
- /* We can understand this */
- break;
- case MS_ADPCM_CODE:
- /* Try to understand this */
- if ( InitMS_ADPCM(format) < 0 ) {
- was_error = 1;
- goto done;
- }
- MS_ADPCM_encoded = 1;
- break;
- case IMA_ADPCM_CODE:
- /* Try to understand this */
- if ( InitIMA_ADPCM(format) < 0 ) {
- was_error = 1;
- goto done;
- }
- IMA_ADPCM_encoded = 1;
- break;
- case MP3_CODE:
- SDL_SetError("MPEG Layer 3 data not supported",
- SDL_SwapLE16(format->encoding));
- was_error = 1;
- goto done;
- default:
- SDL_SetError("Unknown WAVE data format: 0x%.4x",
- SDL_SwapLE16(format->encoding));
- was_error = 1;
- goto done;
- }
- SDL_memset(spec, 0, (sizeof *spec));
- spec->freq = SDL_SwapLE32(format->frequency);
- switch (SDL_SwapLE16(format->bitspersample)) {
- case 4:
- if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) {
- spec->format = AUDIO_S16;
- } else {
- was_error = 1;
- }
- break;
- case 8:
- spec->format = AUDIO_U8;
- break;
- case 16:
- spec->format = AUDIO_S16;
- break;
- default:
- was_error = 1;
- break;
- }
- if ( was_error ) {
- SDL_SetError("Unknown %d-bit PCM data format",
- SDL_SwapLE16(format->bitspersample));
- goto done;
- }
- spec->channels = (Uint8)SDL_SwapLE16(format->channels);
- spec->samples = 4096; /* Good default buffer size */
-
- /* Read the audio data chunk */
- *audio_buf = NULL;
- do {
- if ( *audio_buf != NULL ) {
- SDL_free(*audio_buf);
- }
- lenread = ReadChunk(src, &chunk);
- if ( lenread < 0 ) {
- was_error = 1;
- goto done;
- }
- *audio_len = lenread;
- *audio_buf = chunk.data;
- if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32);
- } while ( chunk.magic != DATA );
- headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */
-
- if ( MS_ADPCM_encoded ) {
- if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) {
- was_error = 1;
- goto done;
- }
- }
- if ( IMA_ADPCM_encoded ) {
- if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) {
- was_error = 1;
- goto done;
- }
- }
-
- /* Don't return a buffer that isn't a multiple of samplesize */
- samplesize = ((spec->format & 0xFF)/8)*spec->channels;
- *audio_len &= ~(samplesize-1);
-
-done:
- if ( format != NULL ) {
- 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 ) {
- spec = NULL;
- }
- 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)
-{
- if ( audio_buf != NULL ) {
- 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 ) {
- SDL_Error(SDL_ENOMEM);
- return(-1);
- }
- if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {
- SDL_Error(SDL_EFREAD);
- SDL_free(chunk->data);
- return(-1);
- }
- return(chunk->length);
-}
generated by cgit v1.1 at 2024-04-27 22:05:00 +0000