NAME
SDL::Audio - SDL Bindings for Audio
CATEGORY
Core, Audio
CONSTANTS
The constants are exported by default. You can avoid this by doing:
use SDL::Audio ();
and access them directly:
SDL::Audio::AUDIO_S16SYS;
or by choosing the export tags below:
Export tag: ':format'
AUDIO_U8
AUDIO_S8
AUDIO_U16LSB
AUDIO_S16LSB
AUDIO_U16MSB
AUDIO_S16MSB
AUDIO_U16
AUDIO_S16
AUDIO_U16SYS
AUDIO_S16SYS
Export tag: ':status'
SDL_AUDIO_STOPPED
SDL_AUDIO_PLAYING
SDL_AUDIO_PAUSED
METHODS
open
use SDL;
use SDL::Audio;
SDL::init(SDL_INIT_AUDIO);
my $desired = SDL::AudioSpec->new();
my $obtained;
SDL::Audio::open( $desired, $obtained );
# $obtained->... (A new SDL::AudioSpec now);
This function opens the audio device with the desired parameters, and returns 0 if successful, placing the actual hardware parameters in the structure pointed to by obtained. If obtained 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.
To open the audio device a desired SDL::AudioSpec must be created.
my $desired = SDL::AudioSpec->new();
You must then fill this structure with your desired audio specifications.
- The desired audio frequency in samples-per-second.
-
$desired->freq
- The desired audio format. See SDL::AudioSpec
-
$desired->format
- The desired channels (1 for mono, 2 for stereo, 4 for surround, 6 for surround with center and lfe).
-
$desired->channels
-
$desired->samples
- This 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 SDL::Audio::lock and SDL::Audio::unlock in your code.
-
THIS IS NOT READY YET
$desired->callback my $callback = sub{ my ($userdata, $stream, $len) = @_; }; $userdata is a reference stored in the userdata field of the SDL::AudioSpec. $stream is a pointer to the audio buffer you want to fill with information and $len is the length of the audio buffer in bytes. $desired->userdata This pointer is passed as the first parameter to the callback function.
SDL::Audio::open reads these fields from the desired SDL::AudioSpec structure passed to the function and attempts to find an audio configuration matching your desired. As mentioned above, if the obtained parameter is NULL then SDL with convert from your desired audio settings to the hardware settings as it plays.
If obtained is NULL then the desired SDL::AudioSpec is your working specification, otherwise the obtained SDL::AudioSpec becomes the working specification and the desired specification can be deleted. The data in the working specification is used when building SDL::AudioCVT's for converting loaded data to the hardware format.
SDL::Audio::open calculates the size and silence fields for both the $desired and $obtained specifications. The size field stores the total size of the audio buffer in bytes, while the silence stores the value used to represent silence in the audio buffer
The audio device starts out playing silence when it's opened, and should be enabled for playing by calling SDL::Audio::pause(0) when you are ready for your audio callback function to be called. Since the audio driver may modify the requested size of the audio buffer, you should allocate any local mixing buffers after you open the audio device.
pause
pause( $bool )
This function pauses and unpauses the audio callback processing. It should be called with $bool = 0
after opening the audio device to start playing sound. This is so you can safely initialize data for your callback function after opening the audio device. Silence will be written to the audio device during the pause.
get_status
int get_status();
Returns either SDL_AUDIO_STOPPED
, SDL_AUDIO_PLAYING
or SDL_AUDIO_PAUSED
depending on the current audio state.
load_wav
SDL::AudioSpec load_wav( $filename, $spec );
This function loads a WAVE file into memory.
If this function succeeds, it returns the given SDL::AudioSpec
, filled with the audio data format of the wave data, and sets buf
to a buffer containing the audio data, and sets len
to the length of that audio buffer, in bytes. You need to free the audio buffer with SDL::Audio::free_wav
when you are done with it.
This function returns NULL and sets the SDL error message if the wave file cannot be opened, uses an unknown data format, or is corrupt. Currently raw, MS-ADPCM and IMA-ADPCM WAVE files are supported.
Example:
use SDL;
use SDL::Audio;
use SDL::AudioSpec;
SDL::init(SDL_INIT_AUDIO);
# Converting some WAV data to hardware format
my $desired = SDL::AudioSpec->new();
my $obtained = SDL::AudioSpec->new();
# Set desired format
$desired->freq(22050);
$desired->channels(1);
$desired->format(AUDIO_S16);
$desired->samples(8192);
# Open the audio device
if( SDL::Audio::open($desired, $obtained) < 0 )
{
printf( STDERR "Couldn't open audio: %s\n", SDL::get_error() );
exit(-1);
}
# Load the test.wav
my $wav_ref = SDL::Audio::load_wav('../../test/data/sample.wav', $obtained);
unless( $wav_ref )
{
warn( "Could not open sample.wav: %s\n", SDL::get_error() );
SDL::Audio::close_audio();
SDL::quit;
exit(-1);
}
my ( $wav_spec, $wav_buf, $wav_len ) = @{$wav_ref};
free_wav
free_wav( $buffer )
After a WAVE file has been opened with load_wav
its data can eventually be freed with free_wav
. buffer
is the buffer created by load_wav
.
convert
SDL::Audio->convert( cvt, data, len )
Converts audio data to a desired audio format.
convert
takes as first parameter cvt
, which was previously initialized. Initializing a SDL::AudioCVT
is a two step process. First of all, the structure must be created via SDL::AudioCVT->build
along with source and destination format parameters. Secondly, the data
and len
fields must be setup. data
should point to the audio data buffer beeing source and destination at once and len
should be set to the buffer length in bytes. Remember, the length of the buffer pointed to by buf should be len*len_mult
bytes in length.
Once the SDL::AudioCVT
structure is initialized, we can pass it to convert
, which will convert the audio data pointed to by data
. If convert
fails undef
is returned, otherwise the converted SDL::AudioCVT
structure.
If the conversion completed successfully then the converted audio data can be read from cvt->buf
. The amount of valid, converted, audio data in the buffer is equal to cvt->len*cvt->len_ratio
.
Example:
use SDL;
use SDL::Audio;
use SDL::AudioSpec;
use SDL::AudioCVT;
SDL::init(SDL_INIT_AUDIO);
# Converting some WAV data to hardware format
my $desired = SDL::AudioSpec->new();
my $obtained = SDL::AudioSpec->new();
# Set desired format
$desired->freq(22050);
$desired->channels(1);
$desired->format(AUDIO_S16);
$desired->samples(8192);
# Open the audio device
if( SDL::Audio::open($desired, $obtained) < 0 )
{
printf( STDERR "Couldn't open audio: %s\n", SDL::get_error() );
exit(-1);
}
# Load the test.wav
my $wav_ref = SDL::Audio::load_wav('../../test/data/sample.wav', $obtained);
unless( $wav_ref )
{
warn( "Could not open sample.wav: %s\n", SDL::get_error() );
SDL::Audio::close_audio();
SDL::quit;
exit(-1);
}
my ( $wav_spec, $wav_buf, $wav_len ) = @{$wav_ref};
# Build AudioCVT
my $wav_cvt = SDL::AudioCVT->build( $wav_spec->format, $wav_spec->channels, $wav_spec->freq,
$obtained->format, $obtained->channels, $obtained->freq);
# Check that the convert was built
if( $wav_cvt == -1 )
{
warn( "Couldn't build converter!\n" );
SDL::Audio::close();
SDL::Audio::free_wav($wav_buf);
SDL::quit();
exit(-1);
}
# And now we're ready to convert
SDL::Audio::convert($wav_cvt, $wav_buf, $wav_len);
# We can freeto original WAV data now
SDL::Audio::free_wav($wav_buf);
TODO: What to do with it? How to use callback? See http://www.libsdl.org/cgi/docwiki.cgi/SDL_ConvertAudio
mix
Mixes audio data
Not implemented yet. See: http://www.libsdl.org/cgi/docwiki.cgi/SDL_MixAudio
lock
lock();
The lock manipulated by these functions protects the callback function. During a lock
period, you can be guaranteed that the callback function is not running. Do not call this from the callback function or you will cause deadlock.
unlock
unlock();
Unlocks a previous lock
call.
close
close();
Shuts down audio processing and closes the audio device.