Compounds | |
| class | syn::AdsrEnvModule |
| ADSR envelope generator. More... | |
| class | syn::DummyOutputModule |
| a synth module that streams the given input into nowhere (eats it). More... | |
| class | syn::FilterModule |
| implements a Source Module concept for AudioIStreams this is basically an IStream -> Module adapter ports available in this module: amplitude (input) mono output (output). More... | |
| class | syn::InstrumentModule |
| an instrument is a group of modules connected in some way. More... | |
| class | syn::IStreamModule |
| implements a Source Module concept for AudioIStreams this is basically an IStream -> Module adapter ports available in this module: amplitude (input) mono output (output). More... | |
| class | syn::MixerModule |
| a synth module that streams the given input into openal. More... | |
| class | syn::OpenALOutputModule |
| a module that streams its input terminal to audio hardware via OpenAL (www.openal.org). More... | |
| class | syn::OperatorModule |
| combines two inputs using an operation. More... | |
| class | syn::OscModuleBase |
| base class for OscModule<type> template type. More... | |
| class | syn::OscModule |
| Source Module for Osc concepts (see syn/Utils/Generator.h for osc) this is basically an Osc -> Module adapter ports available in this module: freq (input) mono audio (output). More... | |
| class | syn::SplitterModule |
| split one input terminal into many. More... | |
| class | syn::WaveTableOscModule |
| implements a Source Module concept for AudioIStreams this is basically an IStream -> Module adapter ports available in this module: amplitude (input) mono output (output). More... | |
| class | syn::Connection |
| connection between two terminals. More... | |
| class | syn::Module |
| a template signal module. More... | |
| class | syn::SampleBuffer |
| a buffer of audio samples. More... | |
| class | syn::SampleBuffer::iterator |
| class | syn::SampleBuffer::frame_iterator |
| class | syn::SampleBufferQueue |
| SampleBuffer Queue. More... | |
| class | syn::SampleBufferRepos |
| repository for sample buffers. More... | |
| class | syn::Terminal |
| A Terminal. More... | |
| class | syn::CSoundOrcInstrumentBuilder |
| class | syn::InstrumentBuilder |
| abstract base class for instrument builder for building compound modules. More... | |
| class | syn::ModuleFactory |
| factory. More... | |
| class | syn::SubsynthInstrumentBuilder |
| Build modules from config files. More... | |
| class | syn::AdsrIStream |
| The ADSR Envelope Generator is a flexible EG that generates a four-stage ADSR voltage envelope that varies from audio_data_traits::min() to audio_data_traits::max() NOTE: ADSR = (A)ttack, (D)elay, (S)ustain, (R)elease. More... | |
| class | syn::AudioContext |
| class | syn::AudioIStream |
| class | syn::AudioOStream |
| Abstract Output stream for audio targets. More... | |
| class | syn::OpenALAudioContext |
| class | syn::OpenALAudioOStream |
| Output stream for OpenAL. More... | |
| class | syn::OpenALSound |
| This simply buffers an entire wav file, and assigns it to an independant source (voice/stream) in openal. More... | |
| class | syn::OscIStream |
| stream for any oscillator (see generator.h for oscillators). More... | |
| class | syn::RawAudioOStream |
| Output stream for writting a .raw audio file... More... | |
| class | syn::SampleBufferIStream |
| class | syn::WavAudioIStream |
| stream for .wav files. More... | |
| class | syn::WavAudioOStream |
| Output stream for writting .wav files... More... | |
| class | syn::WavHeader |
| class | syn::audio_data_traits |
| describes traits for an audio sample i.e. More... | |
| class | syn::audio_data_traits< float > |
| describes traits for an audio sample (float version). More... | |
| class | syn::audio_data_traits< signed char > |
| describes traits for an audio sample (signed char version). More... | |
| class | syn::audio_data_traits< unsigned char > |
| describes traits for an audio sample (unsigned char version). More... | |
| class | syn::audio_data_traits< signed short > |
| describes traits for an audio sample (signed short version). More... | |
| class | syn::audio_data_traits< unsigned short > |
| describes traits for an audio sample (unsigned short version). More... | |
| class | syn::AudioFormat |
| Audio format descriptor this class describes the format of an audio data stream and will be used practically everywhere you pass audio data around. More... | |
| class | syn::FirFilter |
| finite impulse response filter. More... | |
| class | syn::SimpleLowPass |
| class | syn::SimpleHighPass |
| class | syn::SimpleLP |
| class | syn::RbjFilter |
| Robert Bristow-Johnson filter. More... | |
| class | syn::FlyWeightPool |
| FlyWeightPool. More... | |
| class | syn::TriangleOsc |
| non bandlimited triangle generator. More... | |
| class | syn::SineOsc |
| non bandlimited sine/cos generator. More... | |
| class | syn::SquareOsc |
| non bandlimited square generator. More... | |
| class | syn::WhiteNoiseOsc |
| white noise generator. More... | |
| class | syn::PinkNoiseOsc |
| white noise generator. More... | |
| class | syn::AdsrEnvOsc |
| The ADSR Envelope Generator is a flexible EG that generates a four-stage ADSR voltage envelope that varies from 0 to 1. More... | |
| class | syn::MultivariateType |
| class | syn::Singleton |
| You can use this coolio class to make a singleton, just inherit like so... More... | |
| class | syn::WatermarkedQueue |
| repository for sample buffers. More... | |
SampleBufferMethods | |
| SampleBuffer<DATA_TYPE, CHAN>& | clamp (SampleBuffer< DATA_TYPE, CHAN > &buf) |
| clamp data within the valid range really only meaningful for float types... More... | |
| SampleBuffer1f& | operator+= (SampleBuffer1f &lhs, const SampleBuffer1f &rhs) |
| SampleBuffer1f& | operator *= (SampleBuffer1f &lhs, const SampleBuffer1f &rhs) |
Typedefs | |
| typedef OperatorModule<Math::ADDEQUAL> | AddModule |
| Addition operator module. More... | |
| typedef OperatorModule<Math::MULTEQUAL> | MultModule |
| Multiplication operator module. More... | |
| typedef OperatorModule<Math::EQUAL> | CopyModule |
| Assignment operator module. More... | |
| typedef OscModule<TriangleOsc> | TriangleOscModule |
| triangle wave generator. More... | |
| typedef OscModule<WhiteNoiseOsc> | WhiteNoiseOscModule |
| white noise wave generator. More... | |
| typedef OscModule<PinkNoiseOsc> | PinkNoiseOscModule |
| pink noise wave generator. More... | |
| typedef OscModule<SineOsc> | SineOscModule |
| sine wave generator. More... | |
| typedef OscModule<SquareOsc> | SquareOscModule |
| square wave generator. More... | |
| typedef OscModule<AdsrEnvOsc> | AdsrOscModule |
| adsr envelope generator. More... | |
| typedef boost::shared_ptr< Connection> | ConnectionPtr |
| typedef boost::shared_ptr< Module> | ModulePtr |
| typedef SampleBuffer<float, 1> | SampleBuffer1f |
| typedef SampleBuffer<float, 2> | SampleBuffer2f |
| typedef SampleBuffer<signed short, 1> | SampleBuffer1ss |
| typedef SampleBuffer<signed short, 2> | SampleBuffer2ss |
| typedef SampleBuffer<unsigned short, 1> | SampleBuffer1us |
| typedef SampleBuffer<unsigned short, 2> | SampleBuffer2us |
| typedef boost::shared_ptr< SampleBuffer1f> | SampleBufferPtr |
| typedef boost::shared_ptr< SampleBufferQueue> | SampleBufferQueuePtr |
| typedef boost::shared_ptr< Terminal> | TerminalPtr |
| typedef boost::shared_ptr< AudioIStream> | AudioIStreamPtr |
| typedef boost::shared_ptr< AudioOStream> | AudioOStreamPtr |
Enumerations | |
| enum | Endianness { BIG, LITTLE } |
Functions | |
| void | audio_convert (const in &i, out &o) |
| convert one audio sample to a sample of a different format currently works for any type of data supported by audio_data_traits. More... | |
| void | audio_convert (const signed char &i, signed char &o) |
| degenerate cases of audio_convert: types are the same, no conversion needed. More... | |
| void | audio_convert (const unsigned char &i, unsigned char &o) |
| void | audio_convert (const signed short &i, signed short &o) |
| void | audio_convert (const unsigned short &i, unsigned short &o) |
| void | audio_convert (const float &i, float &o) |
| void* | new_audio_data (const AudioFormat &fmt, int samples) |
| create [samples] number of samples of [fmt] formatted audio data this ignores the channels and samp_rate parameters of [fmt]. More... | |
| void | delete_audio_data (void *d) |
| delete the data created by new_audio_data. More... | |
| void | size_calculator (const unsigned int &srcbytes, unsigned int &destbytes) |
| calculate the number of bytes needed to convert srcbytes amount of <in> formated data to <out> format. More... | |
| void | size_calculator (const AudioFormat &srcfmt, AudioFormat destfmt, const unsigned int &srcbytes, unsigned int &destbytes) |
| similar to the templated version above, takes AudioFormat objects instead. More... | |
| void | bytes_calculator (AudioFormat fmt, const unsigned int &samples, unsigned int &destbytes) |
| given a format and num of samples, compute number of bytes needed. More... | |
| bool | audio_channels_convert (const AudioFormat &srcfmt, const AudioFormat &destfmt, const void *srcdata, const int samples, void *destdata, int &destsamples) |
| split or merge the channels in a block of audio data holding [samples] samples returns the number of samples written to dest data. More... | |
| bool | audio_samprate_convert (const AudioFormat &srcfmt, const AudioFormat &destfmt, const void *srcdata, const int samples, void *destdata) |
| resample a signal to a different arbitrary sampling rate. More... | |
| const t* | audio_iterate (const t *data, int x) |
| iterate over the samples in the data. More... | |
| const char* | audio_iterate (const char *data, const AudioFormat &fmt, int x) |
| bool | audio_format_convert (const AudioFormat &srcfmt, const AudioFormat &destfmt, const void *srcdata, const int samples, void *destdata) |
| convert a block of data of one audio format to a block of data in a different format . More... | |
| bool | audio_convert (const AudioFormat &srcfmt, const AudioFormat &destfmt, const void *srcdata, const int samples, void *destdata, int &destsamples) |
| generic signal data convert function. More... | |
| type | audio_clamp (const type &t) |
| clamp a sample to be within audio_data_traits<type>::min() and audio_data_traits<type>::max() this really only makes sense for formats such as FLOAT32, where the data is able to go beyond min and max. More... | |
| void | byteReverse (Type &bytes) |
| void | byteReverse (const syn::Endianness &e, Type &bytes) |
| bool | isLittle () |
| bool | isBig () |
that are audio system independant and simple to use. all handle 32bit floating point formatted samples.
These all conform to the "filter" concept described as follows:
class filterconcept { void filter( float& data ); // changes data void setCutoff( float fc, float sampRate ) }
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Definition at line 40 of file ConnectionPtr.h. Referenced by syn::Terminal::connect().
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Definition at line 40 of file ModulePtr.h. |
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Definition at line 40 of file SampleBufferPtr.h. |
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Definition at line 40 of file SampleBufferQueuePtr.h. |
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Definition at line 40 of file TerminalPtr.h. Referenced by syn::Module::createInput(), and syn::Module::createOutput().
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Definition at line 40 of file AudioIStreamPtr.h. Referenced by syn::WaveTableOscModule::loadFile().
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Definition at line 40 of file AudioOStreamPtr.h. |
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clamp data within the valid range really only meaningful for float types...
Definition at line 248 of file SampleBuffer.h. Referenced by syn::OpenALSound::setPos().
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Definition at line 258 of file SampleBuffer.h. |
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Definition at line 266 of file SampleBuffer.h. |
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convert one audio sample to a sample of a different format currently works for any type of data supported by audio_data_traits.
Definition at line 159 of file AudioFormat.h. Referenced by audio_format_convert(), syn::WavAudioIStream::read(), syn::OscIStream::read(), and syn::OpenALOutputModule::update().
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degenerate cases of audio_convert: types are the same, no conversion needed.
Definition at line 214 of file AudioFormat.h. |
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Definition at line 215 of file AudioFormat.h. |
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Definition at line 216 of file AudioFormat.h. |
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Definition at line 217 of file AudioFormat.h. |
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Definition at line 218 of file AudioFormat.h. |
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create [samples] number of samples of [fmt] formatted audio data this ignores the channels and samp_rate parameters of [fmt].
Definition at line 328 of file AudioFormat.h. |
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delete the data created by new_audio_data.
Definition at line 335 of file AudioFormat.h. |
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calculate the number of bytes needed to convert srcbytes amount of <in> formated data to <out> format.
Definition at line 344 of file AudioFormat.h. |
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similar to the templated version above, takes AudioFormat objects instead.
Definition at line 354 of file AudioFormat.h. |
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given a format and num of samples, compute number of bytes needed.
Definition at line 363 of file AudioFormat.h. Referenced by audio_convert().
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split or merge the channels in a block of audio data holding [samples] samples returns the number of samples written to dest data. you need to allocate destdata. use bytes_calculator(), or new_audio_data() preserves data format, and sampling rate (only affects the channels). Definition at line 373 of file AudioFormat.h. Referenced by audio_convert().
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resample a signal to a different arbitrary sampling rate. preserves number of channels, and data format (only affects samp rate). Definition at line 415 of file AudioFormat.h. |
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iterate over the samples in the data. stride correspondes to number of channels interlaced give a 1 if mono, or if you want to iterate through every sample Definition at line 427 of file AudioFormat.h. |
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Definition at line 432 of file AudioFormat.h. |
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convert a block of data of one audio format to a block of data in a different format . preserves number of channels, and sampling rate (only affects the data format). Definition at line 440 of file AudioFormat.h. Referenced by audio_convert().
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generic signal data convert function. if nessesary, uses audio_format_convert, audio_samprate_convert, audio_channels_convert Definition at line 590 of file AudioFormat.h. |
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clamp a sample to be within audio_data_traits<type>::min() and audio_data_traits<type>::max() this really only makes sense for formats such as FLOAT32, where the data is able to go beyond min and max.
Definition at line 629 of file AudioFormat.h. |
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Definition at line 45 of file Endian.h. Referenced by byteReverse(), and syn::WavAudioIStream::read().
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Definition at line 78 of file Endian.h. Referenced by byteReverse(), and isBig().
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Definition at line 95 of file Endian.h. Referenced by byteReverse(), and syn::WavAudioIStream::read().
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1.2.8.1 written by Dimitri van Heesch,
© 1997-2001