diff options
Diffstat (limited to 'distrib/sdl-1.2.15/src/audio/SDL_wave.c')
| -rw-r--r-- | distrib/sdl-1.2.15/src/audio/SDL_wave.c | 596 |
1 files changed, 596 insertions, 0 deletions
diff --git a/distrib/sdl-1.2.15/src/audio/SDL_wave.c b/distrib/sdl-1.2.15/src/audio/SDL_wave.c new file mode 100644 index 0000000..b4ad6c7 --- /dev/null +++ b/distrib/sdl-1.2.15/src/audio/SDL_wave.c @@ -0,0 +1,596 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 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; + 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 ) { + 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; + + /* 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 ) { + 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); + 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) ); + + /* 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); + *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 */ + + 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); + chunk->data = NULL; + return(-1); + } + return(chunk->length); +} |