diff --git a/project/patches/SDL2/SDL_wave.c b/project/patches/SDL2/SDL_wave.c new file mode 100644 index 0000000..ad0d60f --- /dev/null +++ b/project/patches/SDL2/SDL_wave.c @@ -0,0 +1,624 @@ +/* + Simple DirectMedia Layer + Copyright (C) 1997-2014 Sam Lantinga + + 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, 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; + } + 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]; + + /* 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) { + 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)); + } while ((chunk.magic == FACT) || (chunk.magic == LIST) || (chunk.magic == BEXT)); + + /* Decode the audio data format */ + format = (WaveFMT *) chunk.data; + if (chunk.magic != FMT) { + SDL_SetError("SWIT 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", + 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); + + 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: */ diff --git a/project/patches/SDL2/SDL_wave.h b/project/patches/SDL2/SDL_wave.h new file mode 100644 index 0000000..81efeb4 --- /dev/null +++ b/project/patches/SDL2/SDL_wave.h @@ -0,0 +1,66 @@ +/* + Simple DirectMedia Layer + Copyright (C) 1997-2014 Sam Lantinga + + 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" + +/* WAVE files are little-endian */ + +/*******************************************/ +/* Define values for Microsoft WAVE format */ +/*******************************************/ +#define RIFF 0x46464952 /* "RIFF" */ +#define WAVE 0x45564157 /* "WAVE" */ +#define FACT 0x74636166 /* "fact" */ +#define LIST 0x5453494c /* "LIST" */ +#define FMT 0x20746D66 /* "fmt " */ +#define DATA 0x61746164 /* "data" */ +#define BEXT 0x74786562 /* "bext" */ +#define PCM_CODE 0x0001 +#define MS_ADPCM_CODE 0x0002 +#define IEEE_FLOAT_CODE 0x0003 +#define IMA_ADPCM_CODE 0x0011 +#define MP3_CODE 0x0055 +#define WAVE_MONO 1 +#define WAVE_STEREO 2 + +/* Normally, these three chunks come consecutively in a WAVE file */ +typedef struct WaveFMT +{ +/* Not saved in the chunk we read: + Uint32 FMTchunk; + Uint32 fmtlen; +*/ + Uint16 encoding; + Uint16 channels; /* 1 = mono, 2 = stereo */ + Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */ + Uint32 byterate; /* Average bytes per second */ + Uint16 blockalign; /* Bytes per sample block */ + Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */ +} WaveFMT; + +/* The general chunk found in the WAVE file */ +typedef struct Chunk +{ + Uint32 magic; + Uint32 length; + Uint8 *data; +} Chunk; + +/* vi: set ts=4 sw=4 expandtab: */ diff --git a/tools/build/Build.hx b/tools/build/Build.hx index 6fb4569..adb405c 100644 --- a/tools/build/Build.hx +++ b/tools/build/Build.hx @@ -280,6 +280,8 @@ class Build extends hxcpp.Builder copy("patches/SDL2/SDL_config_windows.h", dir+"/include"); copy("patches/SDL2/SDL_config_linux.h", dir+"/include/SDL_config_minimal.h"); copy("patches/SDL2/SDL_cocoawindow.m", dir+"/src/video/cocoa/SDL_cocoawindow.m"); + copy("patches/SDL2/SDL_wave.c", dir+"/src/audio/SDL_wave.c"); + copy("patches/SDL2/SDL_wave.h", dir+"/src/audio/SDL_wave.h"); //copy("patches/SDL2/SDL_stdinc.h", dir+"/include"); copy("buildfiles/sdl2.xml", dir); runIn(dir, "haxelib", ["run", "hxcpp", "sdl2.xml" ].concat(buildArgs));