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https://github.com/SFML/SFML.git
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0aafc6b6ba
Also moved the commentaries in SFML::Window from the Window_New function to Window_Initialize. git-svn-id: https://sfml.svn.sourceforge.net/svnroot/sfml/branches/sfml2@1746 4e206d99-4929-0410-ac5d-dfc041789085
339 lines
12 KiB
C++
339 lines
12 KiB
C++
/* rbSFML - Copyright (c) 2010 Henrik Valter Vogelius Hansson - groogy@groogy.se
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* This software is provided 'as-is', without any express or
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* implied warranty. In no event will the authors be held
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* liable for any damages arising from the use of this software.
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*
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* Permission is granted to anyone to use this software for any purpose,
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* including commercial applications, and to alter it and redistribute
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* it freely, subject to the following restrictions:
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*
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* 1. The origin of this software must not be misrepresented;
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* you must not claim that you wrote the original software.
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* If you use this software in a product, an acknowledgment
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* in the product documentation would be appreciated but
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* is not required.
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*
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* 2. Altered source versions must be plainly marked as such,
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* and must not be misrepresented as being the original software.
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*
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* 3. This notice may not be removed or altered from any
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* source distribution.
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*/
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#include "SoundBuffer.hpp"
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#include "main.hpp"
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#include <SFML/Audio/SoundBuffer.hpp>
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VALUE globalSoundBufferClass;
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/* Free a heap allocated object
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* Not accessible trough ruby directly!
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*/
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static void SoundBuffer_Free( sf::SoundBuffer *anObject )
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{
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delete anObject;
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}
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/* call-seq:
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* sound_buffer.loadFromFile( filename ) -> true or false
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*
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* Load the sound buffer from a file.
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*
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* Here is a complete list of all the supported audio formats: ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc,
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* ircam, w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
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*/
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static VALUE SoundBuffer_LoadFromFile( VALUE self, VALUE aFileName )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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if( object->LoadFromFile( rb_string_value_cstr( &aFileName ) ) == true )
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{
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return Qtrue;
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}
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else
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{
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return Qfalse;
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}
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}
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/* call-seq:
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* sound_buffer.loadFromSamples( samples, samplesCount, channelsCount, sampleRate ) -> true or false
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*
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* Load the sound buffer from an array of audio samples.
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*
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* The assumed format of the audio samples is 16 bits signed integer.
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*/
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static VALUE SoundBuffer_LoadFromSamples( VALUE self, VALUE someSamples, VALUE aSamplesCount, VALUE aChannelsCount, VALUE aSampleRate )
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{
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const unsigned int rawSamplesCount = FIX2UINT( aSamplesCount );
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const unsigned int rawChannelsCount = FIX2UINT( aChannelsCount );
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const unsigned int rawSampleRate = FIX2UINT( aSampleRate );
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VALIDATE_CLASS( someSamples, rb_cArray, "samples" );
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sf::Int16 * const tempData = new sf::Int16[rawSamplesCount];
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VALUE samples = rb_funcall( someSamples, rb_intern("flatten"), 0 );
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for(unsigned long index = 0; index < rawSamplesCount; index++)
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{
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sf::Int16 val = NUM2INT( rb_ary_entry( samples, index ) );
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tempData[index] = val;
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}
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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bool result = object->LoadFromSamples( tempData, rawSamplesCount, rawChannelsCount, rawSampleRate );
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delete[] tempData;
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if( result == true )
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{
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return Qtrue;
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}
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else
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{
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return Qfalse;
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}
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}
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/* call-seq:
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* sound_buffer.saveToFile( filename ) -> true or false
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*
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* Save the sound buffer to an audio file.
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*
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* Here is a complete list of all the supported audio formats: ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
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* w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
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*/
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static VALUE SoundBuffer_SaveToFile( VALUE self, VALUE aFileName )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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if( object->SaveToFile( rb_string_value_cstr( &aFileName ) ) == true )
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{
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return Qtrue;
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}
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else
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{
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return Qfalse;
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}
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}
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/* call-seq:
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* sound_buffer.getSamples() -> array of samples
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*
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* Get the array of audio samples stored in the buffer.
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*
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* The format of the returned samples is 16 bits signed integer. The total number of samples in this array is given
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* by the getSamplesCount() function.
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*/
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static VALUE SoundBuffer_GetSamples( VALUE self )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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const unsigned int samplesCount = object->GetSamplesCount();
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const sf::Int16 *const samplesPtr = object->GetSamples();
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VALUE samples = rb_ary_new2( samplesCount );
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for(unsigned long index = 0; index < samplesCount; index++)
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{
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rb_ary_store( samples, index, INT2FIX( samplesPtr[index] ) );
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}
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return samples;
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}
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/* call-seq:
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* sound_buffer.getSamplesCount() -> fixnum
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*
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* Get the number of samples stored in the buffer.
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*
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* The array of samples can be accessed with the getSamples() function.
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*/
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static VALUE SoundBuffer_GetSamplesCount( VALUE self )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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return INT2FIX( object->GetSamplesCount() );
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}
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/* call-seq:
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* sound_buffer.getSampleRate() -> fixnum
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*
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* Get the sample rate of the sound.
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*
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* The sample rate is the number of samples played per second. The higher, the better the quality (for example,
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* 44100 samples/s is CD quality).
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*/
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static VALUE SoundBuffer_GetSampleRate( VALUE self )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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return INT2FIX( object->GetSampleRate() );
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}
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/* call-seq:
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* sound_buffer.getChannelsCount() -> float
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*
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* Get the total duration of the sound.
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*/
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static VALUE SoundBuffer_GetChannelsCount( VALUE self )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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return INT2FIX( object->GetChannelsCount() );
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}
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/* call-seq:
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* sound_buffer.getDuration() -> fixnum
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*
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* Get the number of channels used by the sound.
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*
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* If the sound is mono then the number of channels will be 1, 2 for stereo, etc.
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*/
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static VALUE SoundBuffer_GetDuration( VALUE self )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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return rb_float_new( object->GetDuration() );
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}
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/* call-seq:
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* SoundBuffer.new() -> sound_buffer
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* SoundBuffer.new( filename ) -> sound_buffer
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* SoundBuffer.new( samples, samplesCount, channelsCount, sampleRate ) -> sound_buffer
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*
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* Will create a new sound buffer instance.
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*
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* If a filename argument is specified then sound_buffer#loadFromFile will be called on the created instance. If
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* samples, samplesCount, channelsCount and sampleRate are specified then image#loadFromPixels will be called on the
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* created instance.
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*/
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static VALUE SoundBuffer_Initialize( int argc, VALUE *args, VALUE self )
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{
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if( argc > 1 )
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{
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rb_funcall2( self, rb_intern( "loadFromSampels" ), argc, args );
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}
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else if( argc > 0 )
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{
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rb_funcall2( self, rb_intern( "loadFromFile" ), argc, args );
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}
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return self;
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}
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static VALUE SoundBuffer_InitializeCopy( VALUE self, VALUE aSource )
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{
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sf::SoundBuffer *object = NULL;
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Data_Get_Struct( self, sf::SoundBuffer, object );
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sf::SoundBuffer *source = NULL;
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Data_Get_Struct( aSource, sf::SoundBuffer, source );
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*object = *source;
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}
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/* call-seq:
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* SoundBuffer.new() -> sound_buffer
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*
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* Creates an sound buffer instance for us.
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*/
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static VALUE SoundBuffer_New( int argc, VALUE *args, VALUE aKlass )
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{
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sf::SoundBuffer *object = new sf::SoundBuffer();
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VALUE rbData = Data_Wrap_Struct( aKlass, 0, SoundBuffer_Free, object );
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rb_obj_call_init( rbData, argc, args );
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return rbData;
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}
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void Init_SoundBuffer( void )
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{
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/* SFML namespace which contains the classes of this module. */
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VALUE sfml = rb_define_module( "SFML" );
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/* Storage for audio samples defining a sound.
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*
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* A sound buffer holds the data of a sound, which is an array of audio samples.
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*
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* A sample is a 16 bits signed integer that defines the amplitude of the sound at a given time. The sound is then
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* restituted by playing these samples at a high rate (for example, 44100 samples per second is the standard rate used
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* for playing CDs). In short, audio samples are like image pixels, and a SFML::SoundBuffer is similar to a SFML::Image.
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*
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* A sound buffer can be loaded from a file (see loadFromFile() for the complete list of supported formats), from
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* memory or directly from an array of samples. It can also be saved back to a file.
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*
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* Sound buffers alone are not very useful: they hold the audio data but cannot be played. To do so, you need to use
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* the SFML::Sound class, which provides functions to play/pause/stop the sound as well as changing the way it is
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* outputted (volume, pitch, 3D position, ...). This separation allows more flexibility and better performances:
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* indeed a SFML::SoundBuffer is a heavy resource, and any operation on it is slow (often too slow for real-time
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* applications). On the other side, a SFML::Sound is a lightweight object, which can use the audio data of a sound
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* buffer and change the way it is played without actually modifying that data. Note that it is also possible to bind
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* several SFML::Sound instances to the same SFML::SoundBuffer.
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*
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* It is important to note that the SFML::Sound instance doesn't copy the buffer that it uses, it only keeps a reference
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* to it. Thus, a SFML::SoundBuffer must not be destructed while it is used by a SFML::Sound (i.e. never write a function
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* that uses a local SFML::SoundBuffer instance for loading a sound).
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*
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* Usage example:
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*
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* # Declare a new sound buffer
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* buffer = SFML::SoundBuffer.new
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*
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* # Load it from a file
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* if buffer.loadFromFile( "sound.wav" ) == false
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* # error...
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* end
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*
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* # Create a sound source and bind it to the buffer
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* sound1 = SFML::Sound.new
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* sound1.setBuffer( buffer )
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*
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* # Play the sound
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* sound1.play()
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*
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* # Create another sound source bound to the same buffer
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* sound2 = SFML::Sound.new
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* sound2.setBuffer( buffer )
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*
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* # Play it with a higher pitch -- the first sound remains unchanged
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* sound2.setPitch( 2 )
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* sound2.play()
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*/
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globalSoundBufferClass = rb_define_class_under( sfml, "SoundBuffer", rb_cObject );
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// Class methods
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rb_define_singleton_method( globalSoundBufferClass, "new", SoundBuffer_New, -1 );
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// Instance methods
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rb_define_method( globalSoundBufferClass, "initialize", SoundBuffer_Initialize, -1 );
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rb_define_method( globalSoundBufferClass, "initialize_copy", SoundBuffer_InitializeCopy, 1 );
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rb_define_method( globalSoundBufferClass, "loadFromFile", SoundBuffer_LoadFromFile, 1 );
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rb_define_method( globalSoundBufferClass, "loadFromSamples", SoundBuffer_LoadFromSamples, 4 );
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rb_define_method( globalSoundBufferClass, "saveToFile", SoundBuffer_SaveToFile, 1 );
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rb_define_method( globalSoundBufferClass, "getSamples", SoundBuffer_GetSamples, 0 );
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rb_define_method( globalSoundBufferClass, "getSamplesCount", SoundBuffer_GetSamplesCount, 0 );
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rb_define_method( globalSoundBufferClass, "getSampleRate", SoundBuffer_GetSampleRate, 0 );
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rb_define_method( globalSoundBufferClass, "getChannelsCount", SoundBuffer_GetChannelsCount, 0 );
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rb_define_method( globalSoundBufferClass, "getDuration", SoundBuffer_GetDuration, 0 );
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// Instance Aliases
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rb_define_alias( globalSoundBufferClass, "load_from_file", "loadFromFile" );
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rb_define_alias( globalSoundBufferClass, "loadFile", "loadFromFile" );
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rb_define_alias( globalSoundBufferClass, "load_file", "loadFromFile" );
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rb_define_alias( globalSoundBufferClass, "load_from_samples", "loadFromSamples" );
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rb_define_alias( globalSoundBufferClass, "loadSamples", "loadFromSamples" );
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rb_define_alias( globalSoundBufferClass, "load_samples", "loadFromSamples" );
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rb_define_alias( globalSoundBufferClass, "save_to_file", "saveToFile" );
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rb_define_alias( globalSoundBufferClass, "save", "saveToFile" );
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rb_define_alias( globalSoundBufferClass, "get_samples", "getSamples" );
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rb_define_alias( globalSoundBufferClass, "samples", "getSamples" );
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rb_define_alias( globalSoundBufferClass, "get_samples_count", "getSamplesCount" );
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rb_define_alias( globalSoundBufferClass, "samples_count", "getSamplesCount" );
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rb_define_alias( globalSoundBufferClass, "samplesCount", "getSamplesCount" );
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rb_define_alias( globalSoundBufferClass, "get_sample_rate", "getSampleRate" );
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rb_define_alias( globalSoundBufferClass, "sample_rate", "getSampleRate" );
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rb_define_alias( globalSoundBufferClass, "sampleRate", "getSampleRate" );
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rb_define_alias( globalSoundBufferClass, "get_channels_count", "getChannelsCount" );
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rb_define_alias( globalSoundBufferClass, "channels_count", "getChannelsCount" );
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rb_define_alias( globalSoundBufferClass, "channelsCount", "getChannelsCount" );
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rb_define_alias( globalSoundBufferClass, "get_duration", "getDuration" );
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rb_define_alias( globalSoundBufferClass, "duration", "getDuration" );
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}
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