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Diffstat (limited to '3rdparty/SDL2/src/audio/SDL_wave.c')
-rw-r--r--3rdparty/SDL2/src/audio/SDL_wave.c624
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: */