SFML/bindings/ruby/sfml-audio/audio/SoundBuffer.cpp

338 lines
12 KiB
C++

/* rbSFML - Copyright (c) 2010 Henrik Valter Vogelius Hansson - groogy@groogy.se
* This software is provided 'as-is', without any express or
* implied warranty. In no event will the authors be held
* liable for any damages arising from the use of this software.
*
* Permission is granted to anyone to use this software for any purpose,
* including commercial applications, and to alter it and redistribute
* it freely, subject to the following restrictions:
*
* 1. The origin of this software must not be misrepresented;
* you must not claim that you wrote the original software.
* If you use this software in a product, an acknowledgment
* in the product documentation would be appreciated but
* is not required.
*
* 2. Altered source versions must be plainly marked as such,
* and must not be misrepresented as being the original software.
*
* 3. This notice may not be removed or altered from any
* source distribution.
*/
#include "SoundBuffer.hpp"
#include "main.hpp"
#include <SFML/Audio/SoundBuffer.hpp>
VALUE globalSoundBufferClass;
/* Free a heap allocated object
* Not accessible trough ruby directly!
*/
static void SoundBuffer_Free( sf::SoundBuffer *anObject )
{
delete anObject;
}
/* call-seq:
* sound_buffer.loadFromFile( filename ) -> true or false
*
* Load the sound buffer from a file.
*
* Here is a complete list of all the supported audio formats: ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc,
* ircam, w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
*/
static VALUE SoundBuffer_LoadFromFile( VALUE self, VALUE aFileName )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
if( object->LoadFromFile( rb_string_value_cstr( &aFileName ) ) == true )
{
return Qtrue;
}
else
{
return Qfalse;
}
}
/* call-seq:
* sound_buffer.loadFromSamples( samples, samplesCount, channelsCount, sampleRate ) -> true or false
*
* Load the sound buffer from an array of audio samples.
*
* The assumed format of the audio samples is 16 bits signed integer.
*/
static VALUE SoundBuffer_LoadFromSamples( VALUE self, VALUE someSamples, VALUE aSamplesCount, VALUE aChannelsCount, VALUE aSampleRate )
{
const unsigned int rawSamplesCount = FIX2UINT( aSamplesCount );
const unsigned int rawChannelsCount = FIX2UINT( aChannelsCount );
const unsigned int rawSampleRate = FIX2UINT( aSampleRate );
VALIDATE_CLASS( someSamples, rb_cArray, "samples" );
sf::Int16 * const tempData = new sf::Int16[rawSamplesCount];
VALUE samples = rb_funcall( someSamples, rb_intern("flatten"), 0 );
for(unsigned long index = 0; index < rawSamplesCount; index++)
{
sf::Int16 val = NUM2INT( rb_ary_entry( samples, index ) );
tempData[index] = val;
}
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
bool result = object->LoadFromSamples( tempData, rawSamplesCount, rawChannelsCount, rawSampleRate );
delete[] tempData;
if( result == true )
{
return Qtrue;
}
else
{
return Qfalse;
}
}
/* call-seq:
* sound_buffer.saveToFile( filename ) -> true or false
*
* Save the sound buffer to an audio file.
*
* Here is a complete list of all the supported audio formats: ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
* w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
*/
static VALUE SoundBuffer_SaveToFile( VALUE self, VALUE aFileName )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
if( object->SaveToFile( rb_string_value_cstr( &aFileName ) ) == true )
{
return Qtrue;
}
else
{
return Qfalse;
}
}
/* call-seq:
* sound_buffer.getSamples() -> array of samples
*
* Get the array of audio samples stored in the buffer.
*
* The format of the returned samples is 16 bits signed integer. The total number of samples in this array is given
* by the getSamplesCount() function.
*/
static VALUE SoundBuffer_GetSamples( VALUE self )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
const unsigned int samplesCount = object->GetSamplesCount();
const sf::Int16 *const samplesPtr = object->GetSamples();
VALUE samples = rb_ary_new2( samplesCount );
for(unsigned long index = 0; index < samplesCount; index++)
{
rb_ary_store( samples, index, INT2FIX( samplesPtr[index] ) );
}
return samples;
}
/* call-seq:
* sound_buffer.getSamplesCount() -> fixnum
*
* Get the number of samples stored in the buffer.
*
* The array of samples can be accessed with the getSamples() function.
*/
static VALUE SoundBuffer_GetSamplesCount( VALUE self )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
return INT2FIX( object->GetSamplesCount() );
}
/* call-seq:
* sound_buffer.getSampleRate() -> fixnum
*
* Get the sample rate of the sound.
*
* The sample rate is the number of samples played per second. The higher, the better the quality (for example,
* 44100 samples/s is CD quality).
*/
static VALUE SoundBuffer_GetSampleRate( VALUE self )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
return INT2FIX( object->GetSampleRate() );
}
/* call-seq:
* sound_buffer.getChannelsCount() -> float
*
* Get the total duration of the sound.
*/
static VALUE SoundBuffer_GetChannelsCount( VALUE self )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
return INT2FIX( object->GetChannelsCount() );
}
/* call-seq:
* sound_buffer.getDuration() -> fixnum
*
* Get the number of channels used by the sound.
*
* If the sound is mono then the number of channels will be 1, 2 for stereo, etc.
*/
static VALUE SoundBuffer_GetDuration( VALUE self )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
return rb_float_new( object->GetDuration() );
}
/* call-seq:
* SoundBuffer.new() -> sound_buffer
* SoundBuffer.new( filename ) -> sound_buffer
* SoundBuffer.new( samples, samplesCount, channelsCount, sampleRate ) -> sound_buffer
*
* Will create a new sound buffer instance.
*
* If a filename argument is specified then sound_buffer#loadFromFile will be called on the created instance. If
* samples, samplesCount, channelsCount and sampleRate are specified then image#loadFromPixels will be called on the
* created instance.
*/
static VALUE SoundBuffer_Initialize( int argc, VALUE *args, VALUE self )
{
if( argc > 1 )
{
rb_funcall2( self, rb_intern( "loadFromSampels" ), argc, args );
}
else if( argc > 0 )
{
rb_funcall2( self, rb_intern( "loadFromFile" ), argc, args );
}
return self;
}
static VALUE SoundBuffer_InitializeCopy( VALUE self, VALUE aSource )
{
sf::SoundBuffer *object = NULL;
Data_Get_Struct( self, sf::SoundBuffer, object );
sf::SoundBuffer *source = NULL;
Data_Get_Struct( aSource, sf::SoundBuffer, source );
*object = *source;
}
/* call-seq:
* SoundBuffer.new() -> sound_buffer
*
* Creates an sound buffer instance for us.
*/
static VALUE SoundBuffer_Alloc( VALUE aKlass )
{
sf::SoundBuffer *object = new sf::SoundBuffer();
return Data_Wrap_Struct( aKlass, 0, SoundBuffer_Free, object );
}
void Init_SoundBuffer( void )
{
/* SFML namespace which contains the classes of this module. */
VALUE sfml = rb_define_module( "SFML" );
/* Storage for audio samples defining a sound.
*
* A sound buffer holds the data of a sound, which is an array of audio samples.
*
* A sample is a 16 bits signed integer that defines the amplitude of the sound at a given time. The sound is then
* restituted by playing these samples at a high rate (for example, 44100 samples per second is the standard rate used
* for playing CDs). In short, audio samples are like image pixels, and a SFML::SoundBuffer is similar to a SFML::Image.
*
* A sound buffer can be loaded from a file (see loadFromFile() for the complete list of supported formats), from
* memory or directly from an array of samples. It can also be saved back to a file.
*
* Sound buffers alone are not very useful: they hold the audio data but cannot be played. To do so, you need to use
* the SFML::Sound class, which provides functions to play/pause/stop the sound as well as changing the way it is
* outputted (volume, pitch, 3D position, ...). This separation allows more flexibility and better performances:
* indeed a SFML::SoundBuffer is a heavy resource, and any operation on it is slow (often too slow for real-time
* applications). On the other side, a SFML::Sound is a lightweight object, which can use the audio data of a sound
* buffer and change the way it is played without actually modifying that data. Note that it is also possible to bind
* several SFML::Sound instances to the same SFML::SoundBuffer.
*
* It is important to note that the SFML::Sound instance doesn't copy the buffer that it uses, it only keeps a reference
* to it. Thus, a SFML::SoundBuffer must not be destructed while it is used by a SFML::Sound (i.e. never write a function
* that uses a local SFML::SoundBuffer instance for loading a sound).
*
* Usage example:
*
* # Declare a new sound buffer
* buffer = SFML::SoundBuffer.new
*
* # Load it from a file
* if buffer.loadFromFile( "sound.wav" ) == false
* # error...
* end
*
* # Create a sound source and bind it to the buffer
* sound1 = SFML::Sound.new
* sound1.setBuffer( buffer )
*
* # Play the sound
* sound1.play()
*
* # Create another sound source bound to the same buffer
* sound2 = SFML::Sound.new
* sound2.setBuffer( buffer )
*
* # Play it with a higher pitch -- the first sound remains unchanged
* sound2.setPitch( 2 )
* sound2.play()
*/
globalSoundBufferClass = rb_define_class_under( sfml, "SoundBuffer", rb_cObject );
// Class methods
//rb_define_singleton_method( globalSoundBufferClass, "new", SoundBuffer_New, -1 );
rb_define_alloc_func( globalSoundBufferClass, SoundBuffer_Alloc );
// Instance methods
rb_define_method( globalSoundBufferClass, "initialize", SoundBuffer_Initialize, -1 );
rb_define_method( globalSoundBufferClass, "initialize_copy", SoundBuffer_InitializeCopy, 1 );
rb_define_method( globalSoundBufferClass, "loadFromFile", SoundBuffer_LoadFromFile, 1 );
rb_define_method( globalSoundBufferClass, "loadFromSamples", SoundBuffer_LoadFromSamples, 4 );
rb_define_method( globalSoundBufferClass, "saveToFile", SoundBuffer_SaveToFile, 1 );
rb_define_method( globalSoundBufferClass, "getSamples", SoundBuffer_GetSamples, 0 );
rb_define_method( globalSoundBufferClass, "getSamplesCount", SoundBuffer_GetSamplesCount, 0 );
rb_define_method( globalSoundBufferClass, "getSampleRate", SoundBuffer_GetSampleRate, 0 );
rb_define_method( globalSoundBufferClass, "getChannelsCount", SoundBuffer_GetChannelsCount, 0 );
rb_define_method( globalSoundBufferClass, "getDuration", SoundBuffer_GetDuration, 0 );
// Instance Aliases
rb_define_alias( globalSoundBufferClass, "load_from_file", "loadFromFile" );
rb_define_alias( globalSoundBufferClass, "loadFile", "loadFromFile" );
rb_define_alias( globalSoundBufferClass, "load_file", "loadFromFile" );
rb_define_alias( globalSoundBufferClass, "load_from_samples", "loadFromSamples" );
rb_define_alias( globalSoundBufferClass, "loadSamples", "loadFromSamples" );
rb_define_alias( globalSoundBufferClass, "load_samples", "loadFromSamples" );
rb_define_alias( globalSoundBufferClass, "save_to_file", "saveToFile" );
rb_define_alias( globalSoundBufferClass, "save", "saveToFile" );
rb_define_alias( globalSoundBufferClass, "get_samples", "getSamples" );
rb_define_alias( globalSoundBufferClass, "samples", "getSamples" );
rb_define_alias( globalSoundBufferClass, "get_samples_count", "getSamplesCount" );
rb_define_alias( globalSoundBufferClass, "samples_count", "getSamplesCount" );
rb_define_alias( globalSoundBufferClass, "samplesCount", "getSamplesCount" );
rb_define_alias( globalSoundBufferClass, "get_sample_rate", "getSampleRate" );
rb_define_alias( globalSoundBufferClass, "sample_rate", "getSampleRate" );
rb_define_alias( globalSoundBufferClass, "sampleRate", "getSampleRate" );
rb_define_alias( globalSoundBufferClass, "get_channels_count", "getChannelsCount" );
rb_define_alias( globalSoundBufferClass, "channels_count", "getChannelsCount" );
rb_define_alias( globalSoundBufferClass, "channelsCount", "getChannelsCount" );
rb_define_alias( globalSoundBufferClass, "get_duration", "getDuration" );
rb_define_alias( globalSoundBufferClass, "duration", "getDuration" );
}