diff options
Diffstat (limited to '3rdparty/SDL2/src/audio/SDL_wave.c')
-rw-r--r-- | 3rdparty/SDL2/src/audio/SDL_wave.c | 624 |
1 files changed, 0 insertions, 624 deletions
diff --git a/3rdparty/SDL2/src/audio/SDL_wave.c b/3rdparty/SDL2/src/audio/SDL_wave.c deleted file mode 100644 index d173b4a9217..00000000000 --- a/3rdparty/SDL2/src/audio/SDL_wave.c +++ /dev/null @@ -1,624 +0,0 @@ -/* - Simple DirectMedia Layer - Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org> - - This software is provided 'as-is', without any express or implied - warranty. In no event will the authors be held liable for any damages - arising from the use of this software. - - Permission is granted to anyone to use this software for any purpose, - including commercial applications, and to alter it and redistribute it - freely, subject to the following restrictions: - - 1. The origin of this software must not be misrepresented; you must not - claim that you wrote the original software. If you use this software - in a product, an acknowledgment in the product documentation would be - appreciated but is not required. - 2. Altered source versions must be plainly marked as such, and must not be - misrepresented as being the original software. - 3. This notice may not be removed or altered from any source distribution. -*/ -#include "../SDL_internal.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) { - /* const Uint16 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; - Uint8 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) { - return SDL_OutOfMemory(); - } - 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) { - /* const Uint16 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 */ - if (state->index > 88) { - state->index = 88; - } else if (state->index < 0) { - state->index = 0; - } - /* explicit cast to avoid gcc warning about using 'char' as array index */ - step = step_table[(int)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]; - - /* 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 %u channels", - (unsigned int)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) { - return SDL_OutOfMemory(); - } - 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 IEEE_float_encoded, 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; - - SDL_zero(chunk); - - /* 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 { - 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) || (chunk.magic == BEXT) || (chunk.magic == JUNK)); - - /* 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; - } - IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0; - switch (SDL_SwapLE16(format->encoding)) { - case PCM_CODE: - /* We can understand this */ - break; - case IEEE_FLOAT_CODE: - IEEE_float_encoded = 1; - /* 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"); - was_error = 1; - goto done; - default: - SDL_SetError("Unknown WAVE data format: 0x%.4x", - SDL_SwapLE16(format->encoding)); - was_error = 1; - goto done; - } - SDL_zerop(spec); - spec->freq = SDL_SwapLE32(format->frequency); - - if (IEEE_float_encoded) { - if ((SDL_SwapLE16(format->bitspersample)) != 32) { - was_error = 1; - } else { - spec->format = AUDIO_F32; - } - } else { - 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; - case 32: - spec->format = AUDIO_S32; - 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 { - 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 = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels; - *audio_len &= ~(samplesize - 1); - - done: - 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) -{ - 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) { - return SDL_OutOfMemory(); - } - if (SDL_RWread(src, chunk->data, chunk->length, 1) != 1) { - SDL_free(chunk->data); - chunk->data = NULL; - return SDL_Error(SDL_EFREAD); - } - return (chunk->length); -} - -/* vi: set ts=4 sw=4 expandtab: */ |