55 Royalty-Free Audio Tracks for "Methods"

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04:19
I tried out some new gear and methods! more mountainside rain! a complete storm from beginning to end with swelling and then waning rainfall, occasional thunder. Ms stereo bar on mic stand projected through open second-floor window, just under the roof eaves. Recorded july 20th during one of the mid-afternoon brief intense storms we can seem to get frequently in the rocky mountain foothills beginning around june. If you were listening and wondering, the direct sound of the thunder is somewhat occluded as i believe the storm was behind the overhanging roof and house from the perspective of the microphone array. The mountainous terrain and other objects in the field reflected some of that thunderclap, as well as the exterior wall of the home, and so this is all a little bit funky. Mid-side stereo recording:large diaphragm condensers mounted on octavia stereo mic mounting bar:akg perception 220 mid (cardiod) (on top, upside-down)akg perception 400 side (in bidirectional mode) (on bottom, facing left)sound devices mixpre-6 preamp+mixer+recorder (ch1+2 paired to ms stereo, gain @ +21db, fader @ unity, balance at mid/side center) w/48vdc phantom power applied, on-board mid-side encoding and monitoring in l+r, 24/96khz stereo recording. Postprocessing:cooledit 2000: edited to excerpt from longer recording. Normalized recording to 0db. Downsampled to 16bit 48khz. Flac: encoded.
Author: Chromakei
00:00
03:12
I'm attempting to create a controllable thunderstorm for a film, and this is my first legitimate attempt. This recording consists of 4 samples of rain, and another 3 samples of rain+thunder that i recorded one afternoon. Equipment used was the inbuilt mics on a roland r-26, and a sennheiser me66 into a sound devices 702. The clips were recorded at 96khz/24-bit, and they were processed at 48khz/24-bit. For processing, i put the samples into kyma, and crossfaded for texture. The howling wind sound is an analog-style low pass filter's frequency, level, and resonance being controlled by a wacom intuos4 pen/tablet. The rain slowly swells, which was done by changing parameters of a granular reverb. The thunder was also controlled by the wacom tablet, with x, y, and z (pressure) dimensions mapped to making the thunder swell in level, density, and texture. This could have been output in surround, but i don't have that many monitors ;). This style of "rain-synthesis" can also go on indefinitely. Please let me know what you think of the quality of this track; eg, if it sounds real, if the wind sounds ridiculous, too much thunder, etc. Use this sound (wherever) if you want to, or let me know if you'd like an mp3 of this, or for it to last longer. I'd like some credit if you do use it, but it's no big deal. A blog is up explaining the method of creation here:http://www. Kylehughesaudio. Com/2/post/2013/02/tempest. Html.
Author: Tehspaz
00:00
00:21
Noise created by individual oscillators, using audio paint, with different height images, to demonstrate what happens with too few oscillators vs. Plenty. In the end, the result is not random enough to be noise. The first 2-second burst is pure white noise for comparison. Then we have multiple 1-second bursts from audio-paint in a sequence of different image sizes: 50,100,150,200,250,350, 500, 700, 1000, 1350, 1750, 2200, 2700, 3250, 3850, 4500, and 9999 (this corresponds to the number of oscillators). The last burst is longer, and there is 1/2 second gap of silence after the first (reference) burst and before that last (9999) burst. The images other dimension was 20. The spacing of frequencies was exponential, between 40 hz and 18 khz. This is not intended to be useful, just an illustration during a discussion in a forum (http://www. Freesound. Org/forum/sample-requests/35199/?page=2#post75605). As mentioned there, i realized only afterward that exponential spacing would be giving me an approximation to pink noise instead of white noise, so the reference burst at the start is not really a fair comparison. Ideally, i would go back and re-do all this using linear spacing, but that's a lot of trouble. :-) i did, however, change to linear to get a white approximation, but that's a different sound i'll upload separately (c. F. Http://www. Freesound. Org/people/zimbot/sounds/242053/). I don't believe you can get true white noise without at least something being random in your synthesis method.
Author: Zimbot
00:00
00:18
Ringmodulated inversion of my speech from the file https://freesound. Org/people/kb7clx/sounds/648443/ invertedspeechcq. Wav. I took the raw recording and used goldwave's mechanize effect to translate my voice to a center frequency of 14khz. I then demodulated it first at 10. 6 and then 10. 2khz meaning that what comes out is essentially the opposite sideband, offset by 3. 4 and 3. 8khz respectively. 3khz just didn't sound as good. The first i filtered with a low pass of 2. 9khz, the second was filtered to below 3. 4khz to emulate a communications receiver passband. I am speaking upside down as described in this video. Https://www. Youtube. Com/watch?v=q_ykxzcbh-g beginning at 00:03:16. Being blind i can't see their diagram, but i've got my own by ear intuitive method, keeping in mind that oo and ee are farthest from each other, all other vowells get closer the closer they are to the middle of the human voice frequency range. I say: huhlay sue quee, sue quee, sue quee do ux. Cahlloong sue quee sue quee sue quee do ux. The ay in huhllay is like when a spanish speaker says béisbol (baseball). The a in cahlloong is like the a in cat if you're opening wide for the doctor. The oo is like the oo in book. Listen to the other file and you'll hear: hello cq cq cq dx. Calling cq cq cq dx.
Author: Kbclx
00:00
00:01
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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