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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.
Data Fields
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int |
freq |
DSP frequency (samples per second); see Remarks for details |
format |
audio data format; see Remarks for details |
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Uint8 |
channels |
number of channels: 1 mono, 2 stereo; 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 |
a function that is called when the audio device needs more data; see Remarks for details |
void* |
userdata |
a pointer that is passed as the first parameter to callback |
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Code Examples
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Remarks
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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(). We will detail these common members here:
freq specifies the number of samples sent to the sound device every per second. Common values are 11025, 22050 and 44100. The higher the better. green
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format specifies the size and type of each sample element. Values it can take are: Audio format flags may be one of the following values:
Defaults to LSB byte order -or- Basic ??? |
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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 |
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AUDIO_S32 or AUDIO_S32LSB |
32-bit integer samples |
AUDIO_S32MSB |
as above, but big-endian byte order |
float32 support |
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AUDIO_F32 or AUDIO_F32LSB |
32-bit floating point samples |
AUDIO_F32MSB |
as above, but big-endian byte order |
Native audio byte ordering |
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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 |
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channels specifies the number of separate sound channels. 1 is mono (single channel), 2 is stereo (dual channel). green
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. green
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The calculated values silence and size are calculated by SDL_OpenAudio().
Stereo samples are stored in a LRLRLR ordering.
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callback is of the form:
##alt 1st line from header rev 5387?
typedef void (SDLCALL * SDL_AudioCallback)(void* userdata,
##w/ wiki formatting adjustments?
void* SDL_AudioCallback(void* userdata,
##original content below
void callback(void* userdata,
Uint8* stream,
int len);
with the following parameters:
userdata
an application-specific parameter saved in the SDL_AudioSpec structure
stream
a pointer to the audio data buffer filled in by the callback
len
the length of that buffer in bytes
Once the callback returns, the buffer will no longer be valid.
