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view src/audio/SDL_wave.c @ 1133:609c060fd2a2
The MacOSX Carbon/Cocoa/X11 all in one library patch. Relevant emails:
To: SDL Developers <sdl@libsdl.org>
From: =?ISO-8859-1?Q?Anders_F_Bj=F6rklund?= <afb@algonet.se>
Date: Mon, 30 May 2005 23:29:04 +0200
Subject: [SDL] Mac OS X Video Drivers [patch]
I've updated/added the Carbon and X11 video drivers
to the Mac OS X port of SDL 1.2 (the CVS version),
and made the Cocoa driver and runtime *optional*.
The default is still Cocoa, and the "Quartz" driver.
But you can now also use "toolbox" for Carbon, and
"x11" for running with Apple's (or other) X11 server:
export SDL_VIDEODRIVER=x11
export SDL_VIDEO_GL_DRIVER=/usr/X11R6/lib/libGL.dylib
It also checks if the frameworks are available, by a:
#include <Carbon/Carbon.h> or #import <Cocoa/Cocoa.h>
(this should make it configure on plain Darwin as well?)
Here are the new configure targets:
--enable-video-cocoa use Cocoa/Quartz video driver default=yes
--enable-video-carbon use Carbon/QuickDraw video driver default=yes
--enable-video-x11 use X11 video driver default=no
./configure --enable-video-cocoa --enable-video-carbon
--enable-video-x11 \
--x-includes=/usr/X11R6/include --x-libraries=/usr/X11R6/lib
The Carbon version is just an updated version of the old
SDL driver for Mac OS 9, and could probably be improved...
(but it does work, including the Carbon version of SDLmain)
If you disable cocoa, you can run with -framework Carbon only,
and the C version of SDL_main.c. And if you disable carbon too,
you can still use the X11 version which doesn't require SDLmain.
I updated the DrawSprocket version, but did not include it.
(no blitters or VRAM GWorlds etc. available on OS X anyway)
Besides for Mac OS 9, I don't think there's any use for it ?
And note that any performance on Mac OS X equals OpenGL anyway...
You can get "fair" software SDL results on captured CG displays,
but for decent frame rates you need to be using GL for rendering.
Finally, here is the patch itself:
http://www.algonet.se/~afb/SDL-12CVS-macvideo.patch
--anders
PS. It says "video", but as usual it applies to mouse/keyboard too.
------
To: A list for developers using the SDL library <sdl@libsdl.org>
From: =?ISO-8859-1?Q?Anders_F_Bj=F6rklund?= <afb@algonet.se>
Date: Sun, 4 Sep 2005 10:02:15 +0200
Subject: [SDL] Updated Mac patch
Updated the previous Mac patch to disable Carbon by default.
Also "fixed" the SDL.spec again, so that it builds on Darwin.
http://www.algonet.se/~afb/SDL-1.2.9-mac.patch
Also applied fine to SDL12 CVS, when I tried it.
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Thu, 08 Sep 2005 06:16:14 +0000 |
parents | b8d311d90021 |
children | 80f8c94b5199 |
line wrap: on
line source
/* SDL - Simple DirectMedia Layer Copyright (C) 1997-2004 Sam Lantinga This library is free software; you can redistribute it and/or modify it under the terms of the GNU Library General Public License as published by the Free Software Foundation; either version 2 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 Library General Public License for more details. You should have received a copy of the GNU Library General Public License along with this library; if not, write to the Free Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA Sam Lantinga slouken@libsdl.org */ #ifdef SAVE_RCSID static char rcsid = "@(#) $Id$"; #endif #ifndef DISABLE_FILE /* Microsoft WAVE file loading routines */ #include <stdlib.h> #include <string.h> #include "SDL_error.h" #include "SDL_audio.h" #include "SDL_wave.h" #include "SDL_endian.h" #ifndef NELEMS #define NELEMS(array) ((sizeof array)/(sizeof array[0])) #endif 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; 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 = delta; state->iSamp2 = state->iSamp1; state->iSamp1 = 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 *)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; } 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; int c, channels; /* Check to make sure we have enough variables in the state array */ channels = IMA_ADPCM_state.wavefmt.channels; if ( channels > NELEMS(IMA_ADPCM_state.state) ) { SDL_SetError("IMA ADPCM decoder can only handle %d channels", NELEMS(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 *)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? */; } /* Store the initial sample we start with */ decoded[0] = state[c].sample&0xFF; decoded[1] = 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; } 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 MS_ADPCM_encoded, IMA_ADPCM_encoded; int samplesize; /* WAV magic header */ Uint32 RIFFchunk; Uint32 wavelen; Uint32 WAVEmagic; /* 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; } /* 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; } /* Read the audio data format chunk */ chunk.data = NULL; do { if ( chunk.data != NULL ) { free(chunk.data); } lenread = ReadChunk(src, &chunk); if ( lenread < 0 ) { was_error = 1; goto done; } } 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; default: SDL_SetError("Unknown WAVE data format: 0x%.4x", SDL_SwapLE16(format->encoding)); was_error = 1; goto done; } 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 ) { free(*audio_buf); } lenread = ReadChunk(src, &chunk); if ( lenread < 0 ) { was_error = 1; goto done; } *audio_len = lenread; *audio_buf = chunk.data; } while ( chunk.magic != DATA ); 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 ) { free(format); } if ( freesrc && src ) { SDL_RWclose(src); } 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 ) { free(audio_buf); } } static int ReadChunk(SDL_RWops *src, Chunk *chunk) { chunk->magic = SDL_ReadLE32(src); chunk->length = SDL_ReadLE32(src); chunk->data = (Uint8 *)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); free(chunk->data); return(-1); } return(chunk->length); } #endif /* ENABLE_FILE */