Mercurial > sdl-ios-xcode
view docs/man3/SDL_AudioSpec.3 @ 1542:a8bf1aa21020
Fixed bug #15
SDL_blit_A.mmx-speed.patch.txt --
Speed improvements and a bugfix for the current GCC inline mmx
asm code:
- Changed some ops and removed some resulting useless ones.
- Added some instruction parallelism (some gain)
The resulting speed on my Xeon improved upto 35% depending on
the function (measured in fps).
- Fixed a bug where BlitRGBtoRGBSurfaceAlphaMMX() was
setting the alpha component on the destination surfaces (to
opaque-alpha) even when the surface had none.
SDL_blit_A.mmx-msvc.patch.txt --
MSVC mmx intrinsics version of the same GCC asm code.
MSVC compiler tries to parallelize the code and to avoid
register stalls, but does not always do a very good job.
Per-surface blending MSVC functions run quite a bit faster
than their pure-asm counterparts (upto 55% faster for 16bit
ones), but the per-pixel blending runs somewhat slower than asm.
- BlitRGBtoRGBSurfaceAlphaMMX and BlitRGBtoRGBPixelAlphaMMX (and all
variants) can now also handle formats other than (A)RGB8888. Formats
like RGBA8888 and some quite exotic ones are allowed -- like
RAGB8888, or actually anything having channels aligned on 8bit
boundary and full 8bit alpha (for per-pixel alpha blending).
The performance cost of this change is virtually 0 for per-surface
alpha blending (no extra ops inside the loop) and a single non-MMX
op inside the loop for per-pixel blending. In testing, the per-pixel
alpha blending takes a ~2% performance hit, but it still runs much
faster than the current code in CVS. If necessary, a separate function
with this functionality can be made.
This code requires Processor Pack for VC6.
author | Sam Lantinga <slouken@libsdl.org> |
---|---|
date | Wed, 15 Mar 2006 15:39:29 +0000 |
parents | e5bc29de3f0a |
children | 546f7c1eb755 |
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.TH "SDL_AudioSpec" "3" "Tue 11 Sep 2001, 22:58" "SDL" "SDL API Reference" .SH "NAME" SDL_AudioSpec\- Audio Specification Structure .SH "STRUCTURE DEFINITION" .PP .nf \f(CWtypedef struct{ int freq; Uint16 format; Uint8 channels; Uint8 silence; Uint16 samples; Uint32 size; void (*callback)(void *userdata, Uint8 *stream, int len); void *userdata; } SDL_AudioSpec;\fR .fi .PP .SH "STRUCTURE DATA" .TP 20 \fBfreq\fR Audio frequency in samples per second .TP 20 \fBformat\fR Audio data format .TP 20 \fBchannels\fR Number of channels: 1 mono, 2 stereo .TP 20 \fBsilence\fR Audio buffer silence value (calculated) .TP 20 \fBsamples\fR Audio buffer size in samples .TP 20 \fBsize\fR Audio buffer size in bytes (calculated) .TP 20 \fBcallback(\&.\&.)\fR Callback function for filling the audio buffer .TP 20 \fBuserdata\fR Pointer the user data which is passed to the callback function .SH "DESCRIPTION" .PP The \fBSDL_AudioSpec\fR structure is used to describe the format of some audio data\&. This structure is used by \fI\fBSDL_OpenAudio\fP\fR and \fI\fBSDL_LoadWAV\fP\fR\&. While all fields are used by \fBSDL_OpenAudio\fP only \fBfreq\fR, \fBformat\fR, \fBsamples\fR and \fBchannels\fR are used by \fBSDL_LoadWAV\fP\&. We will detail these common members here\&. .TP 20 \fBfreq\fR The number of samples sent to the sound device every second\&. Common values are 11025, 22050 and 44100\&. The higher the better\&. .TP 20 \fBformat\fR Specifies the size and type of each sample element .IP "\fBAUDIO_U8\fP" 10Unsigned 8-bit samples .IP "\fBAUDIO_S8\fP" 10Signed 8-bit samples .IP "\fBAUDIO_U16\fP or \fBAUDIO_U16LSB\fP" 10Unsigned 16-bit little-endian samples .IP "\fBAUDIO_S16\fP or \fBAUDIO_S16LSB\fP" 10Signed 16-bit little-endian samples .IP "\fBAUDIO_U16MSB\fP" 10Unsigned 16-bit big-endian samples .IP "\fBAUDIO_S16MSB\fP" 10Signed 16-bit big-endian samples .IP "\fBAUDIO_U16SYS\fP" 10Either \fBAUDIO_U16LSB\fP or \fBAUDIO_U16MSB\fP depending on you systems endianness .IP "\fBAUDIO_S16SYS\fP" 10Either \fBAUDIO_S16LSB\fP or \fBAUDIO_S16MSB\fP depending on you systems endianness .TP 20 \fBchannels\fR The number of seperate sound channels\&. 1 is mono (single channel), 2 is stereo (dual channel)\&. .TP 20 \fBsamples\fR When used with \fI\fBSDL_OpenAudio\fP\fR this refers to the size of the audio buffer in samples\&. A sample a chunk of audio data of the size specified in \fBformat\fR mulitplied by the number of channels\&. When the \fBSDL_AudioSpec\fR is used with \fI\fBSDL_LoadWAV\fP\fR \fBsamples\fR is set to 4096\&. .SH "SEE ALSO" .PP \fI\fBSDL_OpenAudio\fP\fR, \fI\fBSDL_LoadWAV\fP\fR ...\" created by instant / docbook-to-man, Tue 11 Sep 2001, 22:58