From d8239786b306ffda6d5d73753d01f0ad3358e1a0 Mon Sep 17 00:00:00 2001 From: Jesse Hall Date: Tue, 17 Jul 2012 16:58:55 -0700 Subject: Delete sdl-1.2.12 Change-Id: Ia96f80df04035ae84be3af468c945f2cec14f99c --- distrib/sdl-1.2.12/src/audio/SDL_wave.c | 597 -------------------------------- 1 file changed, 597 deletions(-) delete mode 100644 distrib/sdl-1.2.12/src/audio/SDL_wave.c (limited to 'distrib/sdl-1.2.12/src/audio/SDL_wave.c') 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; iiSamp1 * 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>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; cencoding)) { - 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); -} -- cgit v1.1