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view docs/html/audio.html @ 3880:74e828c64315 SDL-1.2

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Fixed bug #292 I might be on crack here. It looks like SDL_ConvertMono() in src/audio/SDL_audiocvt.c adds the left and right channels of a stereo stream together, and clamps the new mono channel if it would overflow. Shouldn't it be dividing by 2 to average the two sample points instead of clamping? Otherwise the mono sample point's volume doubles in the conversion. This would also make the conversion faster, as it replaces two branches per sample frame with a bitwise shift. --ryan.
author Sam Lantinga <slouken@libsdl.org>
date Sun, 24 Sep 2006 15:45:37 +0000
parents 355632dca928
children
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<HTML
><HEAD
><TITLE
>Audio</TITLE
><META
NAME="GENERATOR"
CONTENT="Modular DocBook HTML Stylesheet Version 1.76b+
"><LINK
REL="HOME"
TITLE="SDL Library Documentation"
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HREF="reference.html"><LINK
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TITLE="SDL_AudioSpec"
HREF="sdlaudiospec.html"><META
NAME="KEYWORD"
CONTENT="audio"><META
NAME="KEYWORD"
CONTENT="function"></HEAD
><BODY
CLASS="CHAPTER"
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><HR
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><DIV
CLASS="CHAPTER"
><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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