Differences between revisions 28 and 29

SDL_AudioSpec

A structure that contains the audio output format. It also contains a callback that is called when the audio device needs more data.

Contents

SDL_AudioSpec

Data Fields

int	freq	DSP frequency (samples per second); see Remarks for details
SDL_AudioFormat	format	audio data format; see Remarks for details
Uint8	channels	number of separate sound channels: see Remarks for details
Uint8	silence	audio buffer silence value (calculated)
Uint16	samples	audio buffer size in samples (power of 2); see Remarks for details
Uint32	size	audio buffer size in bytes (calculated)
SDL_AudioCallback	callback	the function to call when the audio device needs more data; see Remarks for details
void*	userdata	a pointer that is passed to callback (otherwise ignored by SDL)

Code Examples

You can add your code example here

Remarks

This structure is used by SDL_OpenAudioDevice() and SDL_LoadWAV(). While all fields are used by SDL_OpenAudioDevice(), only freq, format, channels, and samples are used by SDL_LoadWAV().

freq specifies the number of sample frames sent to the sound device per second. Common values are 11025, 22050, 44100 and 48000. Larger values produce cleaner audio, in much the same way that larger resolutions produce cleaner graphics.

format specifies the size and type of each sample element and may be one of the following:

8-bit support
AUDIO_S8	signed 8-bit samples
AUDIO_U8	unsigned 8-bit samples
16-bit support
AUDIO_S16LSB	signed 16-bit samples in little-endian byte order
AUDIO_S16MSB	signed 16-bit samples in big-endian byte order
AUDIO_S16SYS	signed 16-bit samples in native byte order
AUDIO_S16	AUDIO_S16LSB
AUDIO_U16LSB	unsigned 16-bit samples in little-endian byte order
AUDIO_U16MSB	unsigned 16-bit samples in big-endian byte order
AUDIO_U16SYS	unsigned 16-bit samples in native byte order
AUDIO_U16	AUDIO_U16LSB
32-bit support (new to SDL 2.0)
AUDIO_S32LSB	32-bit integer samples in little-endian byte order
AUDIO_S32MSB	32-bit integer samples in big-endian byte order
AUDIO_S32SYS	32-bit integer samples in native byte order
AUDIO_S32	AUDIO_S32LSB
float support (new to SDL 2.0)
AUDIO_F32LSB	32-bit floating point samples in little-endian byte order
AUDIO_F32MSB	32-bit floating point samples in big-endian byte order
AUDIO_F32SYS	32-bit floating point samples in native byte order
AUDIO_F32	AUDIO_F32LSB

See SDL_AudioFormat for more info.

channels specifies the number of output channels. As of SDL 2.0, supported values are 1 (mono), 2 (stereo), 4 (quad), and 6 (5.1).

samples specifies a unit of audio data. When used with SDL_OpenAudioDevice() this refers to the size of the audio buffer in sample frames. A sample frame is a chunk of audio data of the size specified in format multiplied by the number of channels. When the SDL_AudioSpec is used with SDL_LoadWAV() samples is set to 4096. This field's value must be a power of two.

The values silence and size are calculated by SDL_OpenAudioDevice().

Channel data is interleaved. Stereo samples are stored in left/right ordering. Quad is stored in front-left/front-right/rear-left/rear-right order. 5.1 is stored in front-left/front-right/center/low-freq/rear-left/rear-right ordering ("low-freq" is the ".1" speaker).

The function prototype for callback is:

void SDL_AudioCallback(void* userdata, Uint8* stream, int len);

where its parameters are:

`userdata`	an application-specific parameter saved in the SDL_AudioSpec structure's userdata field
`stream`	a pointer to the audio data buffer filled in by SDL_AudioCallback()
`len`	the length of that buffer in bytes

Once the callback returns, the buffer will no longer be valid.

The callback must completely initialize the buffer; as of SDL 2.0, this buffer is not initialized before the callback is called. If there is nothing to play, the callback should fill the buffer with silence.

Related Functions

CategoryStruct, CategoryAudio

-  ⇤ ← Revision 28 as of 2011-12-12 19:15:47 → 
  Size: 5165
  Editor: SheenaSmith
  Comment: some sans-Sam editing (shift green to comment)
+   ← Revision 29 as of 2013-08-07 21:22:48 → ⇥
  Size: 3595
  Editor: RyanGordon
  Comment: Rewrote most of this.
-Deletions are marked like this.
+Additions are marked like this.
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-||<tablewidth="100%" style="color: #FF0000;" :> DRAFT||
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+Line 10:
-/* Can '''channels''' only be 1 or 2 or can it handle more? Old wiki lists 4 for surround and 6 for surround w/ center and lfe (sub?). */
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+Line 12:
-||Uint8||'''channels'''||number of separate sound channels: 1 = mono, 2 = stereo||
+||Uint8||'''channels'''||number of separate sound channels: see [[#Remarks|Remarks]] for details||
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+Line 17:
-||void*||'''userdata'''||a pointer that is passed to '''callback'''||
+||void*||'''userdata'''||a pointer that is passed to '''callback''' (otherwise ignored by SDL)||
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+Line 25:
-/* Should the functions [[SDL_LoadWAV_RW]] and [[SDL_OpenAudioDevice]] be added into this remark?  Do they behave like their similarly named counterparts with respect to this struct? */
+This structure is used by [[SDL_OpenAudioDevice]]() and [[SDL_LoadWAV]](). While all fields are used by [[SDL_OpenAudioDevice]](), only '''freq''', '''format''', '''channels''', and '''samples''' are used by [[SDL_LoadWAV]]().
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+Line 27:
-*<<BR>>The [[SDL_AudioSpec]] structure is used to describe the format of some audio data. This structure is used by [[SDL_OpenAudio]]() and [[SDL_LoadWAV]](). While all fields are used by [[SDL_OpenAudio]](), only '''freq''', '''format''', '''channels''', and '''samples''' are used by [[SDL_LoadWAV]]().
+'''freq''' specifies the number of sample frames sent to the sound device per second. Common values are 11025, 22050, 44100 and 48000. Larger values produce cleaner audio, in much the same way that larger resolutions produce cleaner graphics.
-Line 33:
+Line 29:
-'''freq''' specifies the number of samples sent to the sound device per second. Common values are 11025, 22050 and 44100. The higher the better. 
/* I think this info could be moved to Params above and this paragraph removed. */
/* I think "The higher the better." implies that the programmer has a choice and should choose the higher number.  I suspect that the value is actually dictated by the audio data.  Perhaps rewording to say "Higher values produce higher quality sounds/audio output" might be preferable? */
+'''format''' specifies the size and type of each sample element and may be one of the following:
<<Include(SDL_AudioFormat, , , from="=== Audio Format Values ===", to="== Code Examples ==")>>
-Line 37:
+Line 32:
-/* Should this be an include from [[SDL_AudioFormat]] instead? */
'''format''' specifies the size and type of each sample element and may be one of the following:
 ||||''Defaults to LSB byte order'' -or- ''Basic'' ???||
 ||AUDIO_U8||unsigned 8-bit samples.||
 ||AUDIO_S8||signed 8-bit samples.||
 ||AUDIO_U16 or AUDIO_U16LSB||not supported by all hardware (unsigned 16-bit little-endian)||
 ||AUDIO_S16 or AUDIO_S16LSB||not supported by all hardware (signed 16-bit little-endian)||
 ||AUDIO_U16MSB||not supported by all hardware (unsigned 16-bit big-endian)||
 ||AUDIO_S16MSB||not supported by all hardware (signed 16-bit big-endian)||
 ||||''int32 support''||
 ||AUDIO_S32 or AUDIO_S32LSB||32-bit integer samples||
 ||AUDIO_S32MSB||as above, but big-endian byte order||
 ||||''float32 support''||
 ||AUDIO_F32 or AUDIO_F32LSB||32-bit floating point samples||
 ||AUDIO_F32MSB||as above, but big-endian byte order||
 ||||''Native audio byte ordering''||
 ||AUDIO_U16SYS||either AUDIO_U16LSB or AUDIO_U16MSB depending on hardware CPU endianness||
 ||AUDIO_S16SYS||either AUDIO_S16LSB or AUDIO_S16MSB depending on hardware CPU endianness||
 ||AUDIO_S32SYS||either AUDIO_S32LSB or AUDIO_S32MSB depending on hardware CPU endianness||
 ||AUDIO_F32SYS||either AUDIO_F32LSB or AUDIO_F32MSB depending on hardware CPU endianness||
+See [[SDL_AudioFormat]] for more info.
-Line 58:
+Line 34:
+'''channels''' specifies the number of output channels. As of SDL 2.0, supported values are 1 (mono), 2 (stereo), 4 (quad), and 6 (5.1).
-Line 59:
+Line 36:
-/* This list was taken from the header.  A portion was listed in the old wiki as well.  Is it correct that audio change flags and audio flags (as opposed to the above audio format flags) should not be included here? */
+'''samples''' specifies a unit of audio data.  When used with [[SDL_OpenAudioDevice]]() this refers to the size of the audio buffer in sample frames. A sample frame is a chunk of audio data of the size specified in '''format''' multiplied by the number of channels. When the [[SDL_AudioSpec]] is used with [[SDL_LoadWAV]]() '''samples''' is set to 4096. This field's value must be a power of two.
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+Line 38:
-'''samples''' specifies a unit of audio data.  When used with [[SDL_OpenAudio]]() this refers to the size of the audio buffer in samples. A sample is a chunk of audio data of the size specified in '''format''' multiplied by the number of channels. When the [[SDL_AudioSpec]] is used with [[SDL_LoadWAV]]() '''samples''' is set to 4096. 
/* Params above specifies that this is a "power of 2". Would it be appropriate to show how to determine/calculate this value here? */
<<BR>>*
+The values '''silence''' and '''size''' are calculated by [[SDL_OpenAudioDevice]]().
-Line 65:
+Line 40:
-The values '''silence''' and '''size''' are calculated by [[SDL_OpenAudio]]().

Stereo samples are stored in a LRLRLR ordering.
+Channel data is interleaved. Stereo samples are stored in left/right ordering. Quad is stored in front-left/front-right/rear-left/rear-right order. 5.1 is stored in front-left/front-right/center/low-freq/rear-left/rear-right ordering ("low-freq" is the ".1" speaker).
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+Line 44:
- {{{
void SDL_AudioCallback(void*  userdata,
                       Uint8* stream,
                       int    len)
+{{{#!highlight cpp
void SDL_AudioCallback(void* userdata, Uint8* stream, int len);
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+Line 48:
- ||`userdata`||an application-specific parameter saved in the [[SDL_AudioSpec]] structure||
+ ||`userdata`||an application-specific parameter saved in the [[SDL_AudioSpec]] structure's '''userdata''' field||
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+Line 54:
+The callback ''must'' completely initialize the buffer; as of SDL 2.0, this buffer is not initialized before the callback is called. If there is nothing to play, the callback should fill the buffer with silence.
-Line 84:
+Line 57:
- .[[SDL_LoadWAV]] *
+ .[[SDL_LoadWAV]]
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+Line 59:
- .[[SDL_OpenAudio]]

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SDL_AudioSpec

Data Fields

Code Examples

Remarks

Related Functions