74 Royalty-Free Audio Tracks for "Algorithm"

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Stochastic part sound get from http://freesound. Org/people/cbeeching/sounds/116913/. Author cbeeching. The parameters used in the analysis are: window-type = hamming; window-size = 1024; fft-size = 2048; magnitude-threshold = -90; minimum-duration-of-harmonic-tracks = 0. 01; maximum-number-of-harmonics = 100; minimum-fundamental-frequency = 50; maximum-fundamental-frequency = 250; maximum-error-in-f0-detection-algorithm = 7; max-frequency-deviation-in-harmonic-tracks = 0. 01; stochastic approximation-factor = 0. 2.
Author: Manuelrsn
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Harmonic part sound get from http://freesound. Org/people/cbeeching/sounds/116913/. Author cbeeching. The parameters used in the analysis are: window-type = hamming; window-size = 1024; fft-size = 2048; magnitude-threshold = -90; minimum-duration-of-harmonic-tracks = 0. 01; maximum-number-of-harmonics = 100; minimum-fundamental-frequency = 50; maximum-fundamental-frequency = 250; maximum-error-in-f0-detection-algorithm = 7; max-frequency-deviation-in-harmonic-tracks = 0. 01; stochastic approximation-factor = 0. 2.
Author: Manuelrsn
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This is the second part of an audio filtering project (in general, signal processing). This sample shows a noisy recording of my voice, after transforming the signal using the xft (extended fourier transform), applying a filter to cut some frequencies and later applying the inverse xft. All this done with an algorithm in matlab or mathematica written by me. Check the recording before the filtering process by hearing the file "holaoriginal. Wav" and please comment.
Author: Blazeheatnix
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A timbre (rendered with something akin paul naşca’s padsynth algorithm) which has partials of form m + n φ where φ is the golden ratio, and m, n are integers. Though not all the partials, just such a set that this all resembles a usual harmonic timbre. This sequence starts. 1. 000001. 618032. 618033. 618034. 236075. 236075. 854106. 854107. 854108. 47214. This sequence is of several powers of φ up and down from 440 hz (φ⁻² to φ⁵). Look at the spectra for a full dose of xen.
Author: Arseniiv
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Residual output sound obtained by analysis and synthesis of sound speech-female. Wav (http://freesound. Org/people/xserra/sounds/317745/) with harmonic plus residual model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 2019, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 1, maximum number of harmonics = 100, minimum fundamental frequency = 80, maximum fundamental frequency = 300, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01.
Author: Anjds
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Residual output sound obtained by analysis and synthesis of veena concert sound (https://www. Freesound. Org/people/anjds/sounds/377315/ ) with harmonic plus residual model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 1575, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 01, maximum number of harmonics = 100, minimum fundamental frequency = 100, maximum fundamental frequency = 800, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01.
Author: Anjds
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Resynthesized output obtained from the analysis of a sound of erhu : http://freesound. Org/people/ricemutt/sounds/23933/. This was done using the harmonic plus stochastic model implemented in the sms tools: http://mtg. Upf. Edu/technologies/sms. The parameters used for the analysis were:. - window: blackman- window size: 1601- fft size: 2048- magnitude threshold: -90- minimum duration of sinusoidal tracks: 0. 1- maximum number of harmonics: 20- minimum fundamental frequency: 100- maximum fundamental frequency: 2000- maximum error in f0 detection algorithm: 5- max frequency deviation in harmonic tracks: 10stochastic approximation factor: 0. 2.
Author: Wantingchen
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Harmonic sinusoidal output sound obtained by analysis and synthesis of sound speech-female. Wav (http://freesound. Org/people/xserra/sounds/317745/) with harmonic plus stochastic model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 2019, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 1, maximum number of harmonics = 100, minimum fundamental frequency = 80, maximum fundamental frequency = 300, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01,stochastic approximation factor = 0. 7.
Author: Anjds
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Stochastic synthesis output sound obtained by analysis and synthesis of sound speech-female. Wav (http://freesound. Org/people/xserra/sounds/317745/) with harmonic plus stochastic model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 2019, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 1, maximum number of harmonics = 100, minimum fundamental frequency = 80, maximum fundamental frequency = 300, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01,stochastic approximation factor = 0. 7.
Author: Anjds
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Synthesis output sound obtained by analysis and synthesis of sound speech-female. Wav (http://freesound. Org/people/xserra/sounds/317745/) with harmonic plus stochastic model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 2019, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 1, maximum number of harmonics = 100, minimum fundamental frequency = 80, maximum fundamental frequency = 300, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01,stochastic approximation factor = 0. 7.
Author: Anjds
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Stochastic synthesis output sound obtained by analysis and synthesis of veena concert sound (https://www. Freesound. Org/people/anjds/sounds/377315/ ) with harmonic plus stochastic model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 1575, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 01, maximum number of harmonics = 100, minimum fundamental frequency = 100, maximum fundamental frequency = 800, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01,stochastic approximation factor = 0. 2.
Author: Anjds
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Synthesis output sound obtained by analysis and synthesis of veena concert sound (https://www. Freesound. Org/people/anjds/sounds/377315/ ) with harmonic plus stochastic model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 1575, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 01, maximum number of harmonics = 100, minimum fundamental frequency = 100, maximum fundamental frequency = 800, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01,stochastic approximation factor = 0. 2.
Author: Anjds
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Harmonic component obtained from the analysis of a female speech sound: http://www. Freesound. Org/people/xserra/sounds/254374/. This was done using the harmonic plus stochastic model implemented in the sms tools: http://mtg. Upf. Edu/technologies/sms. The parameters used for the analysis were:. - window: blackman- window size: 2001- fft size: 2048- magnitude threshold: -100- minimum duration of sinusoidal tracks: 0. 05- maximum number of harmonics: 100- minimum fundamental frequency: 150- maximum fundamental frequency: 250- maximum error in f0 detection algorithm: 5- max frequency deviation in harmonic tracks: 0. 05stochastic approximation factor: 1.
Author: Wantingchen
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Stochastic component obtained from the analysis of a sound of erhu : http://freesound. Org/people/ricemutt/sounds/23933/. This was done using the harmonic plus stochastic model implemented in the sms tools: http://mtg. Upf. Edu/technologies/sms. The parameters used for the analysis were:. - window: blackman- window size: 1601- fft size: 2048- magnitude threshold: -90- minimum duration of sinusoidal tracks: 0. 1- maximum number of harmonics: 20- minimum fundamental frequency: 100- maximum fundamental frequency: 2000- maximum error in f0 detection algorithm: 5- max frequency deviation in harmonic tracks: 10stochastic approximation factor: 0. 2.
Author: Wantingchen
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Residual signal obtained after subtracting the harmonics from a female speech sound: http://www. Freesound. Org/people/xserra/sounds/254374/. This was done using the harmonic plus residual model implemented in the sms tools: http://mtg. Upf. Edu/technologies/sms. The parameters used for the analysis were:. - window: blackman- window size: 2001- fft size: 2048- magnitude threshold: -100- minimum duration of sinusoidal tracks: 0. 05- maximum number of harmonics: 100- minimum fundamental frequency: 150- maximum fundamental frequency: 250- maximum error in f0 detection algorithm: 5- max frequency deviation in harmonic tracks: 0. 02.
Author: Wantingchen
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Harmonic plus stochastic resynthesis of the female speech sound: http://www. Freesound. Org/people/xserra/sounds/254374/. This was done using the harmonic plus stochastic model implemented in the sms tools: http://mtg. Upf. Edu/technologies/sms. The parameters used for the analysis were:. - window: blackman- window size: 2001- fft size: 2048- magnitude threshold: -100- minimum duration of sinusoidal tracks: 0. 05- maximum number of harmonics: 100- minimum fundamental frequency: 150- maximum fundamental frequency: 250- maximum error in f0 detection algorithm: 5- max frequency deviation in harmonic tracks: 0. 05stochastic approximation factor: 1.
Author: Wantingchen
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Harmonic component obtained from the analysis of a sound of erhu : http://freesound. Org/people/ricemutt/sounds/23933/. This was done using the harmonic plus stochastic model implemented in the sms tools: http://mtg. Upf. Edu/technologies/sms. The parameters used for the analysis were:. - window: blackman- window size: 1601- fft size: 2048- magnitude threshold: -90- minimum duration of sinusoidal tracks: 0. 1- maximum number of harmonics: 20- minimum fundamental frequency: 100- maximum fundamental frequency: 2000- maximum error in f0 detection algorithm: 5- max frequency deviation in harmonic tracks: 10stochastic approximation factor: 0. 2.
Author: Wantingchen
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Harmonic sinusoidal output sound obtained by analysis and synthesis of veena concert sound (https://www. Freesound. Org/people/anjds/sounds/377315/ ) with harmonic plus stochastic model implemented in the sms-tools software package (http://github. Com/mtg/sms-tools)parameters used in analysis are :- window type = blackman, window size(m) = 1575, fft size(n) = 2048, magnitude threshold in db(t) = -100, minimum duration of harmonic tracks = 0. 01, maximum number of harmonics = 100, minimum fundamental frequency = 100, maximum fundamental frequency = 800, maximum error in f0 detection algorithm = 5, max frequency deviation in harmonic tracks = 0. 01,stochastic approximation factor = 0. 2.
Author: Anjds
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Stochastic approximation of the residual signal obtained after subtracting the harmonics from a female speech sound: http://www. Freesound. Org/people/xserra/sounds/254374/. This was done using the harmonic plus stochastic model implemented in the sms tools: http://mtg. Upf. Edu/technologies/sms. The parameters used for the analysis were:. - window: blackman- window size: 2001- fft size: 2048- magnitude threshold: -100- minimum duration of sinusoidal tracks: 0. 05- maximum number of harmonics: 100- minimum fundamental frequency: 150- maximum fundamental frequency: 250- maximum error in f0 detection algorithm: 5- max frequency deviation in harmonic tracks: 0. 05stochastic approximation factor: 1.
Author: Wantingchen
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Piano sound produced with fl studio 10 for the aspma course peer assesment 7 (part 2). Original sound from http://freesound. Org/people/hmartelb/sounds/328547/. This sound is the stochastic component of the hps model analysis using sms-tools. The parameters used in the analysis were:. Window-type = hamming;window-size = 3001;fft-size = 4096;magnitude-threshold = -120;minimum-duration-of-harmonic-tracks = 0. 1;maximum-number-of-harmonics = 200;minimum-fundamental-frequency = 80;maximum-fundamental-frequency = 280;maximum-error-in-f0-detection-algorithm = 5;max-frequency-deviation-in-harmonic-tracks = 0. 01;stochastic approximation-factor = 0. 05;.
Author: Hmartelb
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Sound create use fm synthesis and use 16 bit for high quality. Create sound use c program. Sample have different frequency (power 2) and length. Last several sample have wave frequency above nyquest limit 48000 hz/2 = 24000 hz. Some effect sound like effect in game strider hiryu (ストライダー飛竜) from capcom 1989 (robot gorilla in level siberia) and sonic hedgehog 1 from sega 1990 (sonic jump in large ring at level end if have more than 100 ring). Algorithm:• one carrier and modulate pair• modulate frequency 1. 4 of note (carrier frequency)• modulate amplitude (β) change from 0. 0 to 10. 0 across note• hill envelope with top at 0. 1 of note length.
Author: Sieuamthanh
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Seamless loop of gaspipe or gasleak hissing continuously. This sound is the actual background noise in the frequency range 1200 to 6800 hz from my randomly generated subhorrorambience1. Wav sample. All frequencies below 1 khz (lowend) and above 8 khz (highend) have been filtered out and the volume has been normalized. I release this sound as cc0. You can do whatever you want with it and don't need to mention my name. Tips & tricks for professional mastering of dark ambient loops:. ~ (range) 1200 - 8000 hz, the highmid area of all sounds. ~ (gate) filtering in and adding subtle noise in this frequency range can addatmosphere or "air" to dark ambience but sound noisy like a tape recording. ~ (filter) filtering out noise in this frequency range can make the bass andlower frequencies (20-1000 hz) in dark ambience sound clearer but more dull. ~ (highend) frequencies above 8 khz should always be filtered out completelyand muted to 0 db, since those are only necessary in music and higher sounds.
Author: Zetauri
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We develop iphone app that perform musical analysis on recorded audio from the iphone. Our app implementation make use of the audio queue service to receive raw audio buffers from the audio queue callback. In the first version of our app we had the problem of too much clipping on the recording which degrade the accuracy of our analysis. We also suspected that the noise canceling algorithm in iphone 5 produce distorted sound, which is not much noticeable by human ear but distorted enough to affect our sensitive algorithm. We found that the solution to our problem is to set the audio session mode to kaudiosessionmode_measurement. This session mode is supposed to give maximum freedom for us to control the microphone input, which include turning off the automatic gain control and probably noise canceling as well. The solution works very well except that it introduce a strange waveform pattern in the beginning of all recordings in iphone 5. It is very hard to explain the waveform we get, so i made two recordings at freesound so that you can see it visually. The first recording is made in an almost quite environment, and you can see the weird spike in the beginning of the recording. The second recording (this recording) is made with constant background noise, and you can see that the actual sound wave is offset from the strange curve and gradually increase to its original volume. This waveform only happens on iphone 5 devices that we tested, and there is no problem at all for iphone 4s and older generations. We have tried various settings and the glitch is still unavoidable as long as we set the audio session mode to kaudiosessionmode_measurement. We also find similar glitch in one of our iphone 5 devices, in which the glitch happens even if we try to set just the input gain level without changing the session mode. We are not sure if this is a hardware-related bug in iphone 5, or if it is fixable software glitch in the future version of ios. For the moment we are looking for workaround that can avoid this glitch while automatic gain control and noise canceling are disabled.
Author: Soareschen
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Start sound of mac ii iix iicx iici se/30. Create by dissessemble rom code and use wave table algorithm write c program write wav file. C program below:. /* mac_ii. C *//* boot beep mac ii *//* 2558/09/06 */. #include. #define knumber_samples 30000#define kdelay_note 300#define kwave_table_value 0x30013f10#define ksample_rate 22257 // hz. Void preparewavetable( unsigned short *wavetable, unsigned int value );void updatewavetable( unsigned short *wavetable, unsigned short chiso );void savesound( char *filename, short *sounddata, unsigned int numberframes, unsigned int samplerate );. Int main () {. // ---- wave tableunsigned short wavetable[256];// ---- sound data, stereoshort sounddata[knumber_samples << 1];// ---- increment array (16/16 bit fix point integer)int arrayincrement[] = {3 << 16, 4 << 16, (3 << 16) + 0x2f2, 6 << 16};// ---- prepare wave tablepreparewavetable( wavetable, kwave_table_value );. // ---- array phase (16/16 bit fix point integer)unsigned int arrayphase[] = {0, 0, 0, 0}; // set all = 0. Unsigned int samplenumber = 0;while( samplenumber < knumber_samples ) {. // ---- calculate sampleunsigned int channelleft = 0;unsigned int channelright = 0;unsigned char notenumber = 0;while ( notenumber < 4 ) {// ---- see if should update phase for note, only do if play noteif( samplenumber >= notenumber*kdelay_note ) {// ---- up date phase beforearrayphase[notenumber] += arrayincrement[notenumber];// ---- not let out of range [0; 255]if( arrayphase[notenumber] > 0xff0000 ) // 0xff0000 == 255 << 16arrayphase[notenumber] -= 0xff0000; // return to begin of wave table}unsigned short mauvat = wavetable[arrayphase[notenumber] >> 16];. // ---- add sound componentsif( notenumber < 2 ) // ---- first 2 notes left channelchannelleft += mauvat;else // ---- last 2 notes right channelchannelright += mauvat;// ---- next notenotenumber++;}// ---- save left and right samplessounddata[samplenumber << 1] = (channelleft << 9) - 0x8000; // use << 1 for 16 bitsounddata[(samplenumber << 1) + 1] = (channelright << 9) - 0x8000; // use << 1 for 16 bitupdatewavetable( wavetable, samplenumber & 0xff );samplenumber++;}// ---- save wav filesavesound( "mac ii. Wav", sounddata, samplenumber << 1, ksample_rate ); // multiply 2 because stereo. Return 1;}. Void preparewavetable( unsigned short *wavetable, unsigned int value ) {. // ---- prepare wave tableunsigned short index = 0;unsigned short wavetablevalue = value & 0xff;while( index < 64 ) {wavetable[index] = wavetablevalue; // << 8; // for 16 bitindex++;}. Wavetablevalue = (value >> 8) & 0xff;while( index < 128 ) {wavetable[index] = wavetablevalue; // << 8; // for 16 bitindex++;}. Wavetablevalue = (value >> 16) & 0xff;while( index < 192 ) {wavetable[index] = wavetablevalue; // << 8; // for 16 bitindex++;}wavetablevalue = (value >> 24) & 0xff;while( index < 256 ) {wavetable[index] = wavetablevalue; // << 8; // for 16 bitindex++;}}. Void updatewavetable( unsigned short *wavetable, unsigned short index ) {// ---- get value from wave tableunsigned short value = wavetable[index];// ---- calculate new value for wave tableif( index == 255 ) { // careful at last element of wave tablevalue += wavetable[0];value = (value >> 1);wavetable[0] = value;}else {value += wavetable[index+1];value = (value >> 1);wavetable[index+1] = value;}. }. #pragma mark ---- save wavvoid saveheader( file *filename, unsigned int samplerate );void savesounddatainteger16bit( file *filename, short *sounddata, unsigned int numbersamples );. Void savesound( char *filename, short *sounddata, unsigned int numberframes, unsigned int samplerate ) {// ---- open filefile *file = fopen( filename, "wb" );if( file ) {// ---- "riff"fprintf( file, "riff" );// ---- length sound file - 8unsigned int lengthsoundfile = 32;lengthsoundfile += numberframes << 1; // một không có một mẫu vạt cho kênh trái và phải// ---- save file lengthfputc( (lengthsoundfile) & 0xff, file );fputc( (lengthsoundfile >> 8) & 0xff, file );fputc( (lengthsoundfile >> 16) & 0xff, file );fputc( (lengthsoundfile >> 24) & 0xff, file );// ---- "wave"fprintf( file, "wave" );// ---- save headersaveheader( file, samplerate );// ---- save sound datasavesounddatainteger16bit( file, sounddata, numberframes );// ---- close filefclose( file );}else {printf( "problem save file %s\n", filename );}}. Void saveheader( file *file, unsigned int samplerate ) {// ---- name for header "fmt "fprintf( file, "fmt " );// ---- header lengthfputc( 0x10, file ); // length 16 bytefputc( 0x00, file );fputc( 0x00, file );fputc( 0x00, file );// ---- method for encode, 16 bit pcmfputc( 0x01 & 0xff, file );fputc( (0x00 >> 8) & 0xff, file );// ---- number channels (stereo)fputc( 0x02, file );fputc( 0x00, file );// ---- sample rate (hz)fputc( samplerate & 0xff, file );fputc( (samplerate >> 8) & 0xff, file );fputc( (samplerate >> 16) & 0xff, file );fputc( (samplerate >> 24) & 0xff, file );// ---- number bytes/secondunsigned int numberbytessecond = samplerate << 2; // multiply 4 because short (2 byte) * 2 channelfputc( numberbytessecond & 0xff, file );fputc( (numberbytessecond >> 8) & 0xff, file );fputc( (numberbytessecond >> 16) & 0xff, file );fputc( (numberbytessecond >> 24) & 0xff, file );// ---- byte cho một khung (nên = số lượng mẫu vật * số lượng kênh)// ---- number bytes for sampleunsigned short bytesoneframe = 4; // short (2 byte) * 2 channelunsigned char bitsonesample = 16; // shortfputc( bytesoneframe & 0xff, file );fputc( (bytesoneframe >> 8) & 0xff, file );. Fputc( bitsonesample, file );fputc( 0x00, file );}. Void savesounddatainteger16bit( file *file, short *sounddata, unsigned int numbersamples ) {fprintf( file, "data" );unsigned int datalength = numbersamples << 1; // each sample 2 bytefputc( datalength & 0xff, file );fputc( (datalength >> 8) & 0xff, file );fputc( (datalength >> 16) & 0xff, file );fputc( (datalength >> 24) & 0xff, file );unsigned int sampleindex = 0;while( sampleindex < numbersamples ) {short shortdata = sounddata[sampleindex];fputc( shortdata & 0xff, file );fputc( (shortdata >> 8) & 0xff, file );sampleindex++;}}.
Author: Sieuamthanh
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