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SpasV

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Joined: 2006 (14.4 years ago)    Points: 1,434    Share-Ratio: 67.788    Rank: V.I.P.
Uploaded: 2410210331619 • Downloaded: 35555284059 • no seeds • no downloads
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More about the sound spectra that could be helpful. :-) First of all, what is a sound spectrum? The sound is a wave of pressure we hear when it propagate through the air. A sensor-microphone can generate an electrical signal to mach exactly the pressure wave. This electrical signal I an analog signal meaning it is determined at every moment of a time range and its value is every value in a range of values. The computers don’t process such signals. Instead, they work with discrete time signals. A discrete time signal is determined at discrete moments of time in some time range, so these moments are finite number and its values are taken from a finite set of values, for example the set of all 16-bit binary numbers. Such signals are represented by discrete time functions. We will call the discrete time functions that represent some sound wave digital sound. In order to be played a digital sound needs to be converted back to analog signal. There exist strict mathematical transforms which can transform a function of a time, like a function representing an analog sound signal, to a function of parameter, called frequency. Based on such transform are functions called spectra. Among them is the Power Spectrum or simply spectrum which I’m going to use. The scientific researches have establish that a human can percept sounds whose spectrum parameter is in the range 20 Hz – 22 kHz. According to this the Red Book – the Sony-Philips standard for digital recording of audio CD a digital sound needs to be generated by sampling the analog sound signal at a rate of 44,100 samples per second and the sample digital value needs to be a 16-bit digital number. 44.1 kHz sampling rate has been chosen based on the theory stating that an analog signal can be perfectly reconstructed from an digital time signal if the analog signal is band limited and the sampling rate is at least twice as high as the band limit. There are discrete time transforms to process the discrete time functions and we are using them. Finally, as long as the transforms are strict mathematical operations everything one can conclude from a transform (spectrum) applies to the discrete time function (digital sound) also. In particular, if one conclude that two spectra are close then the digital sound functions are close also. All mp3 spectra shown in this post are obtained using LAME 3.92.4 encoder. Now, there is a spectrum shown on the figure below. It has two views: in a logarithmic scale, which allows for the low frequency range to be observed and in linear scale, which actually hides the low frequency range. The spectrum is calculated based on the Tiesto’s remix of the Delerium [Featuring Sarah McLachlan] – Silence. It is a CD rip. So, you can see the spectrum is band limited with upper limit of 22.05 kHz. This spectrum is of a quality sound: It has almost flat area until frequency 10.5 kHz, then slowly declines until 20 kHz, and there is a transient area 20 – 22 kHz. A worse quality sound usually is below this one. Of course, it depends on the sound itself. Important things to remember: - all audible (20 Hz – 22 kHz) frequencies are present in the spectrum. The spectrum bandwidth is 22 kHz. - There is a simple sound quality rule saying the sound quality is the bandwidth. img Fig 1 Further on I’m going to use only the linear scale representation because the changes I’m going to show are easy visible in the high frequency spectrum range. Next figure shows five spectra: Lossless – it is the original CD rip, Next four are from mp3 (-q0) default encoded at 320 kbps, VBR max quality ( V0), 192 kbps, and 128 kbps digital sounds. img Fig 2 As you can see it’s almost impossible to draw some conclusions based on the differences seen the low and middle frequency range. And I’m not interesting in them. So, further I’m going to show the frequency range I’m going to focus. img Fig 3 The simple bandwidth quality rule here: 320 kbps – 20 kHz, 251 kbps (VBR) – 19 kHz, 192 kbps – 18.5 kHz, 128 kbps – 16.5 kHz. Next, I’m going to discus two sounds: 01-gramatik_-_live_at_the_electric_zoo_(new_york)-sat-09-04-2011-talion.mp3 you can download it from tibalemixes.com 01_gramatik___live_at_the_electric_zoo__new_york__sat_09_04_2011_talion.mp3 you can download through a link provided by mixing.dj Because they both have the same name I’ll rifer them talion and mixing. First, let’s see the talion release. If you take information about the file using MediInfo you’ll get: Encoding settings: -m j -V 0 -q 0 -lowpass 19.5 --vbr-new -b 32 This is exactly the default settings lame is using if you start it this way: lame –q0 –V0. Lame adds the rest. Here is its spectrum along with three more as references. img Fig 4 The talion’s spectrum is below the others obtained from the CD rip. It is different sound indeed. As it is seen talion’s spectrum is band limited at 15 kHz while lame low pass limit imposed by lame’s filter would be 19.5 kHz. It's band limit is 15 kHz and this is because their source is band limited. As I have already said in my previous post, their source is Sirius channel of the satellite radio Sirius XM and it is some kind of FM radio. (Analog FM radios are band limited to 16 kHz.) Because TALiON don’t specify the low pass filter to be used by LAME it uses it's default with a band limit of 19.5 kHz. Because of that it uses 44.1 kHz as a sampling rate, also. And a sample is encoded in average with 216 kbps/44.1 kHz = 4.978 bit per sample. Had they used parameter –lowpass 15 or resample 32, LAME would use 32 kHz as a sampling rate. With such a sampling rate lame vbr encoded Silence (Tiesto remix) has 201 kbps which gives 201/32 = 6.28125 bit per sample in average. Having in main that the source CD rip uses 16 bit/sample it is obvious a 32 kHz sampled encode would be of better quality. (The lower bit per sample introduces so called “quantization noice”.) Besides, this is true for any analog FM radio. They broacast band limited at 16 kHz sound. So, the perfect encode would be sampled at 32.0 kHz. Finally, let’s see the differences between talion and muxing.dj releases. img Fig 5 I would say there are not visible differences. You can see them in the transient range but there they are not important at all. But nevertheless there are differences. The quality of a re-encode, if no special processing has been made, is worse than that of the its losy encoded source. The best re-encode possible would be a lossless encode which will perfectly reconstruct its losy source. But this is meaningless. So, why is that? The reason is as follow. The re-encode is made using a decoded losy encoded source. The chances that the encoder will throw the same information as during the encoding of the original source, even it uses higher bit-rate, is ZERO. That’s way the encoder will throw different information which means it will increase the information lost from the source. So, I don't see any good reason for the TALiON's release to be re-encoded @320 kbps. The result is increased file size and lower sound quality. 
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Hi, I thought to make a post about the sound quality of some sets I have seen here but before that I saw the ignorant comments about my work and I would like to say this for the last time. Sirius radio broadcasts a sound with a spectrum width of 15 kHz. You can easily check this with any TALiON's rip from Electric Zoo Festival New York 09/2011, for example. Just listen to hear and make sure they were broadcasting through Sirius and then run a Spectrum Analysis of the sound file. You'll see a spectrum cut at 15 kHz. The MediaInfo would say about the encoder: Writing library : LAME3.98r Encoding settings : -m j -V 0 -q 0 -lowpass 19.5 --vbr-new -b 32. Then encode ANY CD sound with LAME 3.98 and parameters: -mj -q0 -b95 and check its spectrum out. (-b95 tells encoder to use Constant Bit Rate at 95 kbps.) You'll see the same sound bandwidth of 15 kHz. (Here is what I got: >lame -q0 -mj -b95 airscape.wav LAME 3.98.4 64bits (http://www.mp3dev.org/) CPU features: , SSE (ASM used), SSE2 Resampling: input 44.1 kHz output 32 kHz Using polyphase lowpass filter, transition band: 15097 Hz - 15484 Hz Encoding airscape.wav to airscape.wav.mp3) So, one could conclude Sirius is broadcasting sound filtered at 15 kHz which correspond to a sound quality as if it was mp3 encoded at 96 kbps. BUT as to me, I have been encoding the sound of another satellite channel which former name was The MOVE of XM satellite radio. Sirius was a satellite radio also. Both radios have merged in 2008 and the new satellite radio name is Sirius XM. The former MOVE channel broadcast under the name AREA and is completely different from Sirius division. At least it was using different method of encoding the sound. Beside that, I have reconstructed a CD spectrum of the Area broadcasts using my own filters - programs I have written implementing FFT, if this means anything to you. Finally, I have seen many 320 kbps mp3 rips, even here also, with a spectrum of 15-16 kHz. The LAME encoder filters 320 kbps at 20 kHz and -q0 -V0 (the best variable bit-rate encoding) at 19 kHz. This means those rips have been re-encoded from a band-limited mp3 at 95-128 kbps or FM broadcast source. If you re-encode a lossy encode you'll get an APPROXIMATION to your source and you can get the source quality if you only make a lossless encode. So, it doesn't make any sense to encode an 95 kbps or 128 kbps at 320 kbps only to get the closest mp3 encode to your source. IT IS MUCH BETTER SIMPLY TO RECORD THE SOURCE mp3 STREAM. 
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I am banned on the Comcast network for exceeding this 250 GB/month limit. If you can have FiOS get it. The Verizon people say they do not limit the traffic. Beside, you will pay less for better service. Until then stay under the 250 GB/month limit because on DSL line you'll get 300 kb/s (~40 kB/s) upload. ;-) 
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posted 11.7y ago in Come Celebrate With Pretome!
:-) They do not calculate a ratio. Instead the most important thing is the Hit & Run rule. As long as you are not a Hit & Runner you are fine. The rule applies to any torrent you have downloaded more than 50% of its content. The rule is: You will NOT receive a Hit&Run for a torrent if EITHER seed it for 60 hours OR to ratio 0.75. 60 hours does not have to be contiguous, you can pause your seed and resume later. There are some restrictions and bonuses that encourage seeding as well. As usual, the more you seed the better. 
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posted 11.7y ago in Come Celebrate With Pretome!
img As you may already know, PreToMe is a large General/0Day torrent tracker with 29000+ registered members and over 22000 active torrents. "For this weekend only (Starting July 3rd at 8:00 PM GMT and ending July 5th at midnight GMT) Pretome will have an open sign up to celebrate the birth of our nation!." https://pretome.net/ 
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{quoted:husaberg} State them in the topic tuning your Torrent Client.. and I'll answer. 
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{quoted:luluuu} First, allow the option From a tracker in Options->Connection/Peer Sources. Vuze will connect you to the connectable peers when it receive the information from the tracker - in some (scrape) time interval or if you Update Tracker. Second, if you want me to help you with connectable status contact me in Skype (spas.velev - US) 
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{quoted:husaberg} :-) http://www.bittornado.com/docs/superseed.txt http://wiki.vuze.com/index.php/Super_Seeding http://en.wikipedia.org/wiki/Super-seeding 
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posted 11.7y ago in Still non-connectable?
:-) To get Vuze 4.2 connected to a coonectable peer when in seeding mode you need to set the option: Tools->Options...->Connection Peer Sources: From a tracker With this option set Vuze will initiate a connection to a peer which IP is supplied by the tracker. You need the option Incoming connection also to allow connections initiated by the peers but it obviously works if you are connectable
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posted 11.8y ago in What is this?
:-) I am concerned about so many non-connectable users. UTorrent and Vuze uses one port only. Vuze is the perfect software. The only reason I see is the users, I have shown, were behind a proxy and the tracker determined the IP address by the source address instead by the address sent in the HTTP GET message. If this was a case the proxy could be connectable or not (acctually I don't know). When those users have connected to me Vuze had their real socket addres. 

My Comments



found 81 comments  
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As you can guess I like Josh Wink. :-) But I don't like the ripper TALiON. Here is how they have encoded this mix:lame -m j -V 0 -q 0 -lowpass 19.5. The stupid thing with these parameters is the broadcast was FM because the sound spectrum is 15 kHz width (with a transient range of 1.0 kHz). So, this means the sound can be sampled at 32.0 kHz/s and this means also they have kept 44,1/32.0 = 1.378125 unnecessarily samples than needed! This means also their encode uses 211 kbps/44,1 kHz/s = 4.784 bit/sample. I have re-encoded their encode after having re-sampled and normalized it with -q0 -V0. My result is: get rid of the noise above 15.5 kHz bandwidth thus reducing the unnecessarily encoded noise 202 kbps which means 202/32 = 6.3125 bit/sample compared to their 4.784 bit/sample. 86 MB file size and lame saying "The waveform does not clip and is less than 0.1dB away from full scale" and a sound louder than their's. Unfortunately I cannot correct their mistake. So, if someone can please tell them how to encode a 16 kHz sound stream. Besides, Sirius XM is a FM radio although broadcasted via satellite. 
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added 11.3y ago on John Digweed & M.A.N.D.Y..
:-) 
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added 11.6y ago on Behrouz - Live at Global..
:-) It is always good to have 3 hours of Behrouz. As to the sound quality, here is something that can be helpful. These are spectra of the first hour of the show: (Behrouz - Live @ Pure Behrouz NYC Boat Party - 25-Jul-2009 first hour link provided to me by pozi7ive) Logarithmic scale: http://img256.imageshack.us/i/behrouzlog.png/ Linear scale: http://img194.imageshack.us/i/behrouzlin.png/ img My sound’s spectrum is above and what you see shows it sounds better. Most important is what you can conclude from the “master copy” spectrum. There is a drop at 16 kHz which, in my understanding, is caused by an improper microphone used. After the drop there is a short transition area and the spectrum drops under -104 dB, the digitalization noise level, at 17 kHz. A normal quality mp3 rip at 320 kb/s preserves the spectrum bandwidth up to 20 kHz. Finally, a simple rule of thumb: the wider sound spectrum (up to 22 kHz) the better
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added 11.6y ago on Behrouz - Live at Global..
:-) As to the channel Area 38, you can hardly find worse music radio channel. With its 14 kHz sound bandwidth it is good to be listened in a car maybe, but it is worse than any FM radio channel (bandwidth at least 16 kHz) not to talk about channel Area (XM80), which is NOT Area 38, or my files which are Digital Signal Processed. 
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added 11.6y ago on Behrouz - Live at Global..
:-) Do you think it is a better sound? I doubt. Most probably it is not. 
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:-) 
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{quoted:slash} :-) You are right. I would upload it every week also but Sirius XM radio doesn't broadcast it regularly. Sometimes they broadcast old shows. 
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{quoted:antarez} Of course I remember it. I am not saying an 64 kb/s mp4 stream is the best. But if you decide to upload such a stream your best would be a direct stream ripping. (It is possible to process the sound though and to get better result but an encoder can't do anything better, most probably - worse.) If I were you I would buy a good FM radio antenna in order to have a good signal and record an "in-line" signal. I would buy a 24 bits sound card and I would rip Alpha Radio broadcasts'. The show "120 Minutes with Metropolis" is good. Next super star DJ, most probably, will be John Digweed. There was an exclusive David Dunne's (Had Kandi) show on the radio also. Next, I would have a proper software capable of generating spectrum bandwidth extension in order to extend the 16 kHz FM radio signal spectrum to 22 kHz. Then that would be the best sound you can get from an Alpha Radio broadcast. 
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As long as I know the public sources for the show "120 Minutes with Metropolis" are: 1) An Internet stream from http://eilo.org/120minutes. 2) An FM radio broadcast. ("Alpha-Radio" - broadcasts in Bulgaria.) Personally, knowing the two files originate from Bulgaria, I don't feel proud neither with the first nor with second, rather just opposite. ;-) Edit: :-) To Sasha's fans: for Sasha @ 120 Minutes with Metropolis, Alpha Radio FM (01.07.2009) go to banned link it says "My link is the direct “LINE-IN” from Alpha Radio and no STREAM." 
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:-) Just follow those two rules and you will not make a mistake. If you want to process a file in order to get better quality then learn how to do that. A lossy sound encoder isn't the proper tool for that. 

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