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Date: Mon, 11 Oct 2004 15:17:27 +0300 (EEST) From: Hannu Savolainen Subject: Re: SDL uses obsolete OSS features I did some work on getting OSS to work better with SDL. There have been some problems with select which should be fixed now. I'm having some problems in understanding what is the purpose of the DSP_WaitAudio() routine. I added a return to the very beginning of this routine and commendted out the define for USE_BLOCKING_WRITES. At least lbreakout2 seems to work as well as earlier. The latencies are the same. An ordinary blocking write does exactly the same thing than DSP_WaitAudio does. So I would recommend using the USE_BLOCKING_WRITES approach and removing everything from the DSP_WaitAudio routine. Also enabling USE_BLOCKING_WRITES makes it possible to simplify DSP_PlayAudio() because you don't need to handle the partial writes (the do-while loop). Attached is a patch against SDL-1.2.7. After these changes SDL will use OSS as it's designed to be used (make it as simple as possible). This code should work with all OSS implementations because it uses only the very fundamental features that have been there since the jurassic times.
author Sam Lantinga <slouken@libsdl.org>
date Fri, 12 Nov 2004 21:39:04 +0000
parents 355632dca928
children
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<HTML
><HEAD
><TITLE
>Audio</TITLE
><META
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><DIV
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><H1
><A
NAME="AUDIO"
></A
>Chapter 10. Audio</H1
><DIV
CLASS="TOC"
><DL
><DT
><B
>Table of Contents</B
></DT
><DT
><A
HREF="sdlaudiospec.html"
>SDL_AudioSpec</A
>&nbsp;--&nbsp;Audio Specification Structure</DT
><DT
><A
HREF="sdlopenaudio.html"
>SDL_OpenAudio</A
>&nbsp;--&nbsp;Opens the audio device with the desired parameters.</DT
><DT
><A
HREF="sdlpauseaudio.html"
>SDL_PauseAudio</A
>&nbsp;--&nbsp;Pauses and unpauses the audio callback processing</DT
><DT
><A
HREF="sdlgetaudiostatus.html"
>SDL_GetAudioStatus</A
>&nbsp;--&nbsp;Get the current audio state</DT
><DT
><A
HREF="sdlloadwav.html"
>SDL_LoadWAV</A
>&nbsp;--&nbsp;Load a WAVE file</DT
><DT
><A
HREF="sdlfreewav.html"
>SDL_FreeWAV</A
>&nbsp;--&nbsp;Frees previously opened WAV data</DT
><DT
><A
HREF="sdlaudiocvt.html"
>SDL_AudioCVT</A
>&nbsp;--&nbsp;Audio Conversion Structure</DT
><DT
><A
HREF="sdlbuildaudiocvt.html"
>SDL_BuildAudioCVT</A
>&nbsp;--&nbsp;Initializes a SDL_AudioCVT structure for conversion</DT
><DT
><A
HREF="sdlconvertaudio.html"
>SDL_ConvertAudio</A
>&nbsp;--&nbsp;Convert audio data to a desired audio format.</DT
><DT
><A
HREF="sdlmixaudio.html"
>SDL_MixAudio</A
>&nbsp;--&nbsp;Mix audio data</DT
><DT
><A
HREF="sdllockaudio.html"
>SDL_LockAudio</A
>&nbsp;--&nbsp;Lock out the callback function</DT
><DT
><A
HREF="sdlunlockaudio.html"
>SDL_UnlockAudio</A
>&nbsp;--&nbsp;Unlock the callback function</DT
><DT
><A
HREF="sdlcloseaudio.html"
>SDL_CloseAudio</A
>&nbsp;--&nbsp;Shuts down audio processing and closes the audio device.</DT
></DL
></DIV
><P
>Sound on the computer is translated from waves that you hear into a series of 
values, or samples, each representing the amplitude of the wave.  When these
samples are sent in a stream to a sound card, an approximation of the original
wave can be recreated.  The more bits used to represent the amplitude, and the
greater frequency these samples are gathered, the closer the approximated
sound is to the original, and the better the quality of sound.</P
><P
>This library supports both 8 and 16 bit signed and unsigned sound samples,
at frequencies ranging from 11025 Hz to 44100 Hz, depending on the 
underlying hardware.  If the hardware doesn't support the desired audio
format or frequency, it can be emulated if desired (See 
<A
HREF="sdlopenaudio.html"
><TT
CLASS="FUNCTION"
>SDL_OpenAudio()</TT
></A
>)</P
><P
>A commonly supported audio format is 16 bits per sample at 22050 Hz.</P
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