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view src/audio/SDL_wave.c @ 1192:54aa9aa32327
To: sdl@libsdl.org
From: Christian Walther <cwalther@gmx.ch>
Date: Fri, 18 Nov 2005 23:39:02 +0100
Subject: [SDL] Mouse position bugs on Mac OS X
The attached patch fixes a few bugs in SDL related to the mouse position
in windowed mode on Mac OS X, reproduced using the attached minimal test
program - at least here on 10.3.9, with SDL CVS from today. Could anyone
test whether the bugs exist and are fixed by the patch on 10.2 and 10.4?
1. When using OpenGL, the vertical mouse positions obtained through
events or SDL_GetMouseState() are off by one.
2. When using OpenGL, SDL_WarpMouse() inverts the y coordinate.
3. Clicks on the topmost pixel row of the window are not recognized.
1 and 2 do not occur in non-OpenGL mode, while 3 does. All three only
occur in windowed mode, not in fullscreen.
The cause for 1 and 3 is that in Cocoa, "the location of the mouse"
seems to be defined as "the location of the top left corner of the mouse
pointer's hot pixel" (this is not documented, it's just what I found out
here), which together with the fact that Cocoa's usual y coordinates
start at the bottom and increase upwards means that the y coordinate of
the mouse runs from 1 to h, not from 0 to h-1, in a window of height h.
If it does work on 10.2 and 10.4 (I'll try to test it as soon as I can,
but at the moment all I have at hand is 10.3.9), can this be applied to
the CVS?
-Christian
To: sdl@libsdl.org
From: Christian Walther <cwalther@gmx.ch>
Date: Mon, 28 Nov 2005 10:41:51 +0100
Subject: [SDL] Re: Mouse position bugs on Mac OS X
I wrote:
> I'll try to test it as soon as I can, but at the moment all I have at hand is 10.3.9
So, here are the results of my tests (with patched and unpatched
frameworks compiled with Xcode 1.5 (gcc 3.3) on 10.3.9):
On 10.1.5, my test program doesn't run because of "Undefined symbols:
SDL undefined reference to _CGMainDisplayID expected to be defined in
Carbon". I guess not supporting 10.1 was a deliberate decision then and
that's OK with me.
On 10.2.8, 10.3.9, and 10.4.0, the bugs exist as described in my
original post and are fixed by my patch. That is, there is no difference
between pre/post 10.3 and the patched version works correctly in all
combinations of GL/non-GL and windowed/fullscreen.
I therefore recommend the patch for inclusion.
-Christian
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Mon, 28 Nov 2005 13:58:26 +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 */