view docs/man3/SDL_OpenAudio.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
line wrap: on
line source

.TH "SDL_OpenAudio" "3" "Tue 11 Sep 2001, 22:58" "SDL" "SDL API Reference" 
.SH "NAME"
SDL_OpenAudio\- Opens the audio device with the desired parameters\&.
.SH "SYNOPSIS"
.PP
\fB#include "SDL\&.h"
.sp
\fBint \fBSDL_OpenAudio\fP\fR(\fBSDL_AudioSpec *desired, SDL_AudioSpec *obtained\fR);
.SH "DESCRIPTION"
.PP
This function opens the audio device with the \fBdesired\fR parameters, and returns 0 if successful, placing the actual hardware parameters in the structure pointed to by \fBobtained\fR\&. If \fBobtained\fR is NULL, the audio data passed to the callback function will be guaranteed to be in the requested format, and will be automatically converted to the hardware audio format if necessary\&. This function returns -1 if it failed to open the audio device, or couldn\&'t set up the audio thread\&.
.PP
To open the audio device a \fBdesired\fR \fI\fBSDL_AudioSpec\fR\fR must be created\&. 
.PP
.nf
\f(CWSDL_AudioSpec *desired;
\&.
\&.
desired=(SDL_AudioSpec *)malloc(sizeof(SDL_AudioSpec));\fR
.fi
.PP
 You must then fill this structure with your desired audio specifications\&.
.IP "\fBdesired\fR->\fBfreq\fR" 10The desired audio frequency in samples-per-second\&.
.IP "\fBdesired\fR->\fBformat\fR" 10The desired audio format (see \fI\fBSDL_AudioSpec\fR\fR)
.IP "\fBdesired\fR->\fBsamples\fR" 10The desired size of the audio buffer in samples\&. This number should be a power of two, and may be adjusted by the audio driver to a value more suitable for the hardware\&. Good values seem to range between 512 and 8192 inclusive, depending on the application and CPU speed\&. Smaller values yield faster response time, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time\&. A stereo sample consists of both right and left channels in LR ordering\&. Note that the number of samples is directly related to time by the following formula: ms = (samples*1000)/freq
.IP "\fBdesired\fR->\fBcallback\fR" 10This should be set to a function that will be called when the audio device is ready for more data\&. It is passed a pointer to the audio buffer, and the length in bytes of the audio buffer\&. This function usually runs in a separate thread, and so you should protect data structures that it accesses by calling \fI\fBSDL_LockAudio\fP\fR and \fI\fBSDL_UnlockAudio\fP\fR in your code\&. The callback prototype is: 
.PP
.nf
\f(CWvoid callback(void *userdata, Uint8 *stream, int len);\fR
.fi
.PP
 \fBuserdata\fR is the pointer stored in \fBuserdata\fR field of the \fBSDL_AudioSpec\fR\&. \fBstream\fR is a pointer to the audio buffer you want to fill with information and \fBlen\fR is the length of the audio buffer in bytes\&.
.IP "\fBdesired\fR->\fBuserdata\fR" 10This pointer is passed as the first parameter to the \fBcallback\fP function\&.
.PP
\fBSDL_OpenAudio\fP reads these fields from the \fBdesired\fR \fBSDL_AudioSpec\fR structure pass to the function and attempts to find an audio configuration matching your \fBdesired\fR\&. As mentioned above, if the \fBobtained\fR parameter is \fBNULL\fP then SDL with convert from your \fBdesired\fR audio settings to the hardware settings as it plays\&.
.PP
If \fBobtained\fR is \fBNULL\fP then the \fBdesired\fR \fBSDL_AudioSpec\fR is your working specification, otherwise the \fBobtained\fR \fBSDL_AudioSpec\fR becomes the working specification and the \fBdesirec\fR specification can be deleted\&. The data in the working specification is used when building \fBSDL_AudioCVT\fR\&'s for converting loaded data to the hardware format\&.
.PP
\fBSDL_OpenAudio\fP calculates the \fBsize\fR and \fBsilence\fR fields for both the \fBdesired\fR and \fBobtained\fR specifications\&. The \fBsize\fR field stores the total size of the audio buffer in bytes, while the \fBsilence\fR stores the value used to represent silence in the audio buffer
.PP
The audio device starts out playing \fBsilence\fR when it\&'s opened, and should be enabled for playing by calling \fI\fBSDL_PauseAudio\fP(\fB0\fR)\fR when you are ready for your audio \fBcallback\fR function to be called\&. Since the audio driver may modify the requested \fBsize\fR of the audio buffer, you should allocate any local mixing buffers after you open the audio device\&.
.SH "EXAMPLES"
.PP
.nf
\f(CW/* Prototype of our callback function */
void my_audio_callback(void *userdata, Uint8 *stream, int len);

/* Open the audio device */
SDL_AudioSpec *desired, *obtained;
SDL_AudioSpec *hardware_spec;

/* Allocate a desired SDL_AudioSpec */
desired=(SDL_AudioSpec *)malloc(sizeof(SDL_AudioSpec));

/* Allocate space for the obtained SDL_AudioSpec */
obtained=(SDL_AudioSpec *)malloc(sizeof(SDL_AudioSpec));

/* 22050Hz - FM Radio quality */
desired->freq=22050;

/* 16-bit signed audio */
desired->format=AUDIO_S16LSB;

/* Mono */
desired->channels=0;

/* Large audio buffer reduces risk of dropouts but increases response time */
desired->samples=8192;

/* Our callback function */
desired->callback=my_audio_callback;

desired->userdata=NULL;

/* Open the audio device */
if ( SDL_OpenAudio(desired, obtained) < 0 ){
  fprintf(stderr, "Couldn\&'t open audio: %s
", SDL_GetError());
  exit(-1);
}
/* desired spec is no longer needed */
free(desired);
hardware_spec=obtained;
\&.
\&.
/* Prepare callback for playing */
\&.
\&.
\&.
/* Start playing */
SDL_PauseAudio(0);\fR
.fi
.PP
.SH "SEE ALSO"
.PP
\fI\fBSDL_AudioSpec\fP\fR, \fI\fBSDL_LockAudio\fP\fR, \fI\fBSDL_UnlockAudio\fP\fR, \fI\fBSDL_PauseAudio\fP\fR
...\" created by instant / docbook-to-man, Tue 11 Sep 2001, 22:58