Mercurial > sdl-ios-xcode
diff src/audio/SDL_wave.c @ 1895:c121d94672cb
SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
author | Sam Lantinga <slouken@libsdl.org> |
---|---|
date | Mon, 10 Jul 2006 21:04:37 +0000 |
parents | 7995cc87b777 |
children | 6abc7e6f9817 |
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line diff
--- a/src/audio/SDL_wave.c Thu Jul 06 18:01:37 2006 +0000 +++ b/src/audio/SDL_wave.c Mon Jul 10 21:04:37 2006 +0000 @@ -27,571 +27,592 @@ #include "SDL_wave.h" -static int ReadChunk(SDL_RWops *src, Chunk *chunk); +static int ReadChunk(SDL_RWops * src, Chunk * chunk); -struct MS_ADPCM_decodestate { - Uint8 hPredictor; - Uint16 iDelta; - Sint16 iSamp1; - Sint16 iSamp2; +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]; +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) +static int +InitMS_ADPCM(WaveFMT * format) { - Uint8 *rogue_feel; - Uint16 extra_info; - int i; + 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); + /* 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) +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; + 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); + 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) +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; + 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; + /* 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]; + /* 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; - } + /* 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; + /* 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; + 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); + ++encoded; + samplesleft -= 2; + } + encoded_len -= MS_ADPCM_state.wavefmt.blockalign; + } + SDL_free(freeable); + return (0); } -struct IMA_ADPCM_decodestate { - Sint32 sample; - Sint8 index; +struct IMA_ADPCM_decodestate +{ + Sint32 sample; + Sint8 index; }; -static struct IMA_ADPCM_decoder { - WaveFMT wavefmt; - Uint16 wSamplesPerBlock; - /* * * */ - struct IMA_ADPCM_decodestate state[2]; +static struct IMA_ADPCM_decoder +{ + WaveFMT wavefmt; + Uint16 wSamplesPerBlock; + /* * * */ + struct IMA_ADPCM_decodestate state[2]; } IMA_ADPCM_state; -static int InitIMA_ADPCM(WaveFMT *format) +static int +InitIMA_ADPCM(WaveFMT * format) { - Uint8 *rogue_feel; - Uint16 extra_info; + 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); + /* 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) +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; + 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; + /* 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; - } + /* 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); + /* 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) +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; + 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; + 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; + 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; - } + ++encoded; + } } -static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) +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; + 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; + /* 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; + /* 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? */; - } + /* 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; - } + /* 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); + /* 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) +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; + 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; + /* WAV magic header */ + Uint32 RIFFchunk; + Uint32 wavelen = 0; + Uint32 WAVEmagic; + Uint32 headerDiff = 0; - /* FMT chunk */ - WaveFMT *format = NULL; + /* 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; + } - /* 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 */ + /* 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) ); + /* 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 */ + /* 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 */ + /* 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; - } - } + 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); + /* 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); + 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) +void +SDL_FreeWAV(Uint8 * audio_buf) { - if ( audio_buf != NULL ) { - SDL_free(audio_buf); - } + if (audio_buf != NULL) { + SDL_free(audio_buf); + } } -static int ReadChunk(SDL_RWops *src, Chunk *chunk) +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); + 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); } + +/* vi: set ts=4 sw=4 expandtab: */