SDL::Audio (3)
Leading comments
Automatically generated by Pod::Man 4.09 (Pod::Simple 3.35) Standard preamble: ========================================================================
NAME
SDL::Audio - SDL Bindings for AudioCATEGORY
Core, AudioCONSTANTS
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
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
- The desired size of the audio buffer in samples. This number should be a power of two, and may be adjusted by the audio driver to a value more suitable for the hardware. Good values seem to range between 512 and 8192 inclusive, depending on the application and CPUspeed. Smaller values yield faster response time, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time. A stereo sample consists of both right and left channels inLRordering. Note that the number of samples is directly related to time by the following formula: ms = (samples*1000)/freq
-
$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
If obtained is
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
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
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
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 being 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, $wav_len); # We can free original WAV data now SDL::Audio::free_wav($wav_buf);
mix
Mixes audio dataNot implemented yet. See: <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.