# My Digital Audio Computer



## Gregr (Nov 2, 2010)

I have consistently overlooked my computer and I have not included this as part of my equipment list. Well I am not sure why but finally here is my computer:

MoBo - MSI K9A2 Platinum
CPU - AMD 5200+ Duel core
RAM - 6GB of Kingston HyperX DDR2 
Video Card - Asus 4850 with 512MB of GDDR3 (I think)
Sound Card - HT Omega Claro+ 24Bit/192khz capable
Windows 7 Professional 64Bit
Web Browser - Firefox
Audio - 24Bit/192khz from "Pandora One" $36 Annually for commercial free 24/192 streaming

Other Programs I like:

Adobe Photoshop CS2
Canon EOS Utility programs for the EOS DSLR 30D and 20D
Win Media Center for organizing my Audio Files
BobCAD/CAM for programming CNC Router

:sweat:


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## Chester (Feb 19, 2007)

http://www.pandora.com/pandora_one

"When listening on the web, experience 192K bits per second audio. More bits mean better sounding music."

it is 192 kbits/sec, not 192 khz  not sure if its 24 bit or not, since it is compressed...

nice setup tho!


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## gorb (Sep 5, 2010)

I'm pretty sure it's just 16/44 192kbps mp3.


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## Gregr (Nov 2, 2010)

OK Men, Chester & gorb,

Why do I feel like the lamb being led to slaughter? Because in a round about sort of way the numbers and their meaning are reversed quite easily. It seems if I feel like it I can say 96kHz 16Bit and 192kbps may just be a new lossless compression for PCM on DVD no, no, on Blu-Ray..., could be, I just do not know.

There is so much about the digital realm I am yet to learn. Memory, digital memory is cheap and fast, this is a great combination. It means a computer can accomplish very complex tasks in most cases, immediately. You.ve heard of computers solving mathematical quandaries in less than an hour that would have taken the sharpest mathematical mind a lifetime to solve. Anyway..., fast and memory.

Then somebody came up with the idea of Modulating Pulse Code (or pulse code modulation PCM)or taking the sine wave of a sound breaking it up into equal segments using some form of modulation that is used in both constructing a profile and deconstructing same. Or analog to digital conversion (ADC) and/or digital to analog conversion (DAC).

As As PCM (pulse code mod) progressed first on CD's with 16Bit/44.?kHz what they were doing is measuring 16 bits of depth..., OK here is where I might loose myself

OK..., in order for me to be understood I need to write something that both of us understand. We need many things we both agree are meaningful. The Pulse code consists of Bit Depth and time or Frequency. I know you know about frequency but in this case we are concerned with the number of cycles per second. in the example above we are looking at 44.? thousand cycles per second (very Fast). That is, the PCM is cycling and/or sampling (making a notation) 44.? times per second what is the PCM sampling..., it is sampling an analog sound wave 16 bits deep. 

So for each cycle we get a string of 16 digits (bits) that is a representation of an analog sound wave or multiple sound waves and harmonies and drum and cymbal. 
Don't forget we are talking about the digital world where everything we commonly know is represented by a string of binary numbers in a computer. the each letter of the alphabet and many other characters are represented by 8 bits it takes 8 bits on binary info to create a letter like "C"..., looking something like this "01000110" (don't quote me). But I do know those 8 bits of info make up 1Byte. and so every letter, number and special charecter on the modern computer keyboard is made up of 8bits or 1Byte.

New back to music. With music we've added time to the equation. meaning we have broken a second into 44.? thousand segments and in each segment we have.................., 16bits "0011010010100010" and each of these groups of numbers is part of a musical note, group of notes, a harmony of instruments, etc., etc., 

So it is 24bit/192kHz of PCM. Take a look at this. This might help solidify these ideas and give you an image of what I have tried to describe. But we have come a long way and the numbers you mention could be some form of new algorithm and or compression. I don't have the answer to that.

I will say though I have read evidence of very knowledgeable people believing ar correctly talking about kHz as if it were an audible signal. When in this case it is only a part of a representation of a signal

Enough is Enough. For today then............................................................................... !!

Let me know what this does for you. :sweat:

Thanks 

Greg lddude:


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## Gregr (Nov 2, 2010)

Sorry about the spelling I didn't reread until I sent the note.

But here I also forgot to send this info from Wikipedia:

Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form for digital audio in computers and various Blu-ray, Compact Disc and DVD formats, as well as other uses such as digital telephone systems. A PCM stream is a digital representation of an analog signal, in which the magnitude of the analogue signal is sampled regularly at uniform intervals, with each sample being quantized to the nearest value within a range of digital steps.

PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that each sample can take.

Google PCM

Greg


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## gorb (Sep 5, 2010)

I have no idea what you are trying to say. Regardless, Pandora One = 192kbps. I don't know if they are using mp3/aac/ogg/wma though, but it is probably mp3.

http://blog.pandora.com/faq/contents/64.html


> High Quality 192Kbps audio, on your computer only:
> 
> on our website www.pandora.com and
> when using the Pandora One Desktop Application.


I do not like how they keep saying "High Quality 192kbps audio." Lossy is never high quality. If they were streaming flac that would be different.


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## Gregr (Nov 2, 2010)

Hello gorb,:wave:

Thank you for the link. The 192Kbps they are referring to is bandwidth. Forget the name if you want but what it means is this..., Pandora is downloading to you at a rate of 192Kbps when you use 
"Pandora One". However if you want to listen to Pandora One, Pandora recommends your computer be capable of downloading 300Kbps. For the free Pandora, Pandora recommends a minimum of 150Kbps download ability. 

Bandwidth has nothing to do recording in 24bit/192kHz except that it fit inside of the 192Kbps bandwidth.

Here is a little more info and three sites you can test your computer to see if you can download @ 300Kbps: I'll explain the 300Kbps further but at another time.


You need a broadband connection with consistent bandwidth of at least 150 Kbps, e.g. DSL or Cable Internet, in order to run Pandora.

(Dial-up and ISDN connections are not supported. Satellite Internet connections may be problematic, especially if you have a dial-up "upstream" connection.)


NOTE: If you have upgraded to a Pandora One subscription account and wish to use the High Quality audio streams, then we recommend having at least 300 Kbps of bandwidth. Note that you have the option to select Normal Quality audio on any computers you use in locations where limited bandwidth is an issue.


If you're not sure of the speed of your internet connection, you can test your connection speed at any of these sites:

* http://www.speedtest.net
* http://www.dslreports.com/stest
* http://us.mcafee.com/root/speedometer :sweat:


IMPORTANT NOTE: The minimum bandwidths listed above are recommended for typical usage of Pandora, such as checking email or surfing the web while listening to Pandora. On the other hand, if you intend to listen to Pandora during other types of heavy internet use-- e.g. downloading large files, streaming video, and so on-- then you will need additional bandwidth to ensure best results. 

Good Luck 

Greg





you need to be able to download files :yikes


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## gorb (Sep 5, 2010)

Yes, 192kbps is the size of the data stream because the file itself is encoded at a rate of 192 kbps. Any modern connection will be able to handle that stream without issue.


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## gorb (Sep 5, 2010)

[email protected] said:


> Hi Phillip,
> 
> Pandora on the Web or the Desktop app plays 64k AAC+ for free listeners and 192k for PandoraOne subscribers.
> 
> ...


PandoraOne = 192kbps AAC+
regular Pandora (website or desktop app) = 64kbps AAC+
In home devices = 128kbps MP3
mobile devices = maximum quality of 64kbps AAC+, can be less


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## Chester (Feb 19, 2007)

I was not trying to 'pick' on you or anything; I just didn't think 192 khz sounded correct, so I looked it up: it turned out that it was not correct, so I 'put it out there' (what the correct information is) so that others would not read it and then be uninformed - also, then you would be more informed.

It was merely a mix up between sample rate and the bit rate of an audio stream.

In a digital audio file, there are 3 metrics that largely impact the sound.
Bit rate
Sample Rate
Bit depth

the bit rate is how much data per second the computer is reading/receiving in order to play the audio; the bitrate of an audio cd (uncompressed) is 1411.2 kbit/s; the sample rate of a FLAC file (losslessly compressed audio) tends to be between 700-1100 kbit/s in my experience; the bitrate of mp3's can get up to 320 kbit/s ... divide any of these numbers by 8 and that will give you how many kilobytes per second of data you have

sample rate is how many 'slices' per second of a sound source you have or another way to word that: how many samples of voltage per second you have: typical audio cd's have a sample rate of 44.1 khz, dividing this by 2 tells you the theoretical maximum frequency you can reproduce with a given sample rate (i.e. a CD can in theory play a 22050 hz frequency... kind of, there are other issues at frequencies very close to half the sample rate (which is called the Nyquist frequency if you are interested)

bit depth is how many bits per sample there are, which determines the dynamic range of the sound; if you have 16 bits (as you do in a CD) you have 2^16 = 65536 different 'levels' or steps of voltage that can be represented for each sample... in compressed audio the bit depth does not 'exist' technically (at least in mp3, and as far as I know MOST lossy forms of audio compression)

something you may have noticed: 16 (bits per sample) * 44100 (samples per second) *2 (channels) = 1411200 /1000 = 1411.2 kbits/s (which is where I started when explaining bit rate, sample rate, and bit depth)

please note that this only applies for uncompressed audio, barring psycho-acoustic encoding and high pass filters which are reducing the amount of data that needs to be saved; we can also reverse what I did above and show: 192 (kbit/s) *1000 *1/44100 (take out sample rate) *1/2 (take out channels) =~ 2.17687... or a little more than 2 bits of bit depth... which I find interesting (I never calculated that out till just now)

anyways, I hope that explains things a bit more: I also do agree with gorb when he says that lossy is never high quality... there are many reasons for that: though you may not notice the quality difference (I do)


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## Gregr (Nov 2, 2010)

Hi Chester,

Please do not interpret any of the following as indicative of your wrong about anything. I am not a complete expert and all of this digital coding is complex and convoluted and easily misinterpreted for many reasons. 
I like the work you are doing and I am very interested to read all of the material you've organized and written here for all to see. I always see just a little clearer after reading someone's knowledgeable reiteration of text material. It always reads better with a practical point of reference.

My take on digital is only about music. How music is trans-coded into a digital representation of the original creation. I stand by my description of my audio card and Pandora's streaming 24bit/192kHz data. Here's why. 

Your letter from Pandora was very helpful. I was wondering what standards they were using for downloads because as you and I know it is not just music keeping Pandora going or that encoding to CD or DVD is not just 16bit/44.1 kHz all the way up to 24bit/192kHz. I knew there was more but didn't know where to begin.
Well you have been of tremendous help to me especially in this area.

The following is from Wikipedia:

Audio on a DVD-Audio disc can be stored in many different bit depth/sampling rate/channel combinations:

16-, 20- or 24-bit depth

44.1 kHz 48 kHz 88.2 kHz 96 kHz 176.4 kHz 192 kHz
_______________________________________________________
Mono (1.0) Yes Yes Yes Yes Yes Yes
Stereo (2.0) Yes Yes Yes Yes Yes Yes
Stereo (2.1) Yes Yes Yes Yes No No
Stereo + mono surround (3.0 or 3.1) Yes Yes Yes Yes No No
Quad (4.0 or 4.1) Yes Yes Yes Yes No No
3-stereo (3.0 or 3.1) Yes Yes Yes Yes No No
3-stereo + mono surround (4.0 or 4.1) Yes Yes Yes Yes No No
Full surround (5.0 or 5.1) Yes Yes Yes Yes No No

Different bit depth/sampling rate/channel combinations can be used on a single disc. For instance, a DVD-Audio disc may contain a 96 kHz/24-bit 5.1-channel audio track as well as a 192 kHz/24-bit stereo audio track. Also, the channels of a track can be split into two groups stored at different resolutions. For 
example, the front speakers could be 96/24, while the surrounds are 48/20.


Audio is stored on the disc in Linear PCM format, which is either uncompressed or losslessly compressed with Meridian Lossless Packing.[1] The maximum permissible total bit rate is 9.6 Megabits per second. Channel/resolution combinations that would exceed this need to be compressed. In uncompressed modes, it is possible to get up to 96/16 or 48/24 in 5.1, and 192/24 in stereo. To store 5.1 tracks in 88.2/20, 88.2/24, 96/20 or 96/24 MLP encoding is mandatory.

The above is a partial copy of a Wikipedia page.

The AAC Plus that Pandora uses is the same as HE-ACC+. Within AAC+ are many potential profiles because of reverse compatibility with MPEG-1, MPEG-2 which gave birth to MP3 and MPEG-4 and MP4 Audio is also part of the AAC+ along with psycho-acoustical info etc etc etc. I do not have all of the Data, spec's and other numbers. 

From where I sit digital Audio is encoded using 16bit depth/44.1kHz sampling rate = 706kbps of bandwidth this seems to me as where we have missed what the other is saying.
I have the formula actually several for ACC+. There are too many contributers to AAC+ for it to be simple as a standard meeting ISO and IEC Reg's.

Thanks for your patience 

Greg


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## Gregr (Nov 2, 2010)

I just reread

YES 16bits of depth X 44.1kHz sampling per second X 2 Channels = 1412 kilobitspersecond (kbps) or 176kilobytespersecond (kBps) of download speed (broadband) necessary for most MP3 CD's. Does this include compression?

Greg


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## Gregr (Nov 2, 2010)

Chester,

I just reread again. You had me questioning again. Each time I question myself it just help silidify a very complex and convoluted subject. And , I know I don't have the "big picture " yet. 
But bit depth is what is sampled for each cycle of electric current or Audio eventually.

Here is another segment of Wikipedia:

In digital audio, bit depth describes the number of bits of information recorded for each sample. Bit depth directly corresponds to the resolution of each sample in a set of digital audio data. Common examples of bit depth include CD quality audio, which is recorded at 16 bits, and DVD-Audio, which can support up to 24-bit audio.

Thanks

Greg


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## Chester (Feb 19, 2007)

soo... you understand now? (thats what I think your last post was trying to say  )

if so, I am glad you figured it out, you are correct in stating that it is a complex subject; especially depending on your background regarding computers/electronics


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## Gregr (Nov 2, 2010)

Chester,

I wonder how closely you have read anything I wrote to you. What I have written is only in support of my original statements. I made no error. I stand by everything I have written. In fact I wonder about your seemingly mixing of terms. 

Thanks for the challenge 

Greg


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## Chester (Feb 19, 2007)

"In fact I wonder about your seemingly mixing of terms." please quote to me where I have mixed two terms, I do not know to what you are referring.

When I was discussing 1441.2 kbits/s I was referring to WAV audio files (i.e. uncompressed sound) that would be found on a commercial audio CD (99.9% of cds you would find in a store), I also explained where that number comes from (44100*16*2).

I mentioned FLAC audio, which is losslessly compressed (think a zip file, you can get back the original when you decompress it).

This math does not apply to mp3's, mp3 compression maxes out at a bit rate rate of 320 kbit/s: at a bit rate of 192 kbit/s, I calculated that on average, there are 2.17 bits of audio per sample assuming a 44100 khz sample rate; however by the nature of how audio compression works, this is not 'really' the case, just an interesting comparison (in my opinion).

keep in mind (and I think this may be where the confusion lies) that in digital audio one can choose any arbitrary number for the sample rate and bit depth (in uncompressed audio, the bit rate is as I have explained, derived by multiplying the two numbers, so you cannot 'pick' your bit rate unless you do not care about one of the other two values.)

Your statement that your pandora is playing 24 bit/192khz audio is correct in so far as that may be what the sample rate and bit depth is that your computer is sending to your stereo; however this is irrelevant because that is not the entire signal chain; the signal chain is 'starting' with pandora, which is sourced from cds (44.1khz, 16 bit) then compressed (AAC+ 192kbp/s), then sent to you and decompressed, then re-sampled by your computer at 192khz 24 bits, then sent to your stereo.

The thing is, the 'worst part of the chain' is the quality you will actually get, you cannot record a voicemail of yourself (8khz sample rate for phone communications I believe), email it to yourself, play it on your computer, and tell me that it sounds as though it is '192khz and 24 bit audio', while yes, the voice mail you recorded may be being represented by that many samples per second when it is sent to your stereo receiver and it may be being represented by 24 bits, you are basically oversampling by 24x (192/8), which does not actually create any *new* data (higher frequencies) for you to hear (yes, there is aliasing artifacts from re sampling the audio however that is an artifact and was not part of the original signal which is your voice in the physical world.)

Also, the original bit depth will be determining the noise floor. I do not know what bit-depth a phone call uses, for arguments sake let's say 8 bits (its probably less, and they do compress it too, but I am talking about uncompressed audio), you will never get more than 8 bits of dynamic range (loudest to quietest) out of a source signal that is 8 bits of bit depth, anything 'quieter' than that is already lost and has been quantized (rounded) to the smallest value you could represent in the 8 bit system.

It is my assertion that listening to pandora (compressed audio) on your computer is effectively listening to a voice mail that you are playing on your computer (compressed audio), while the pandora music is using better compression and a higher data rate (yielding a better sounding result), either way, you are listening to compressed audio which is being re-sampled to a rate of 192khz and 24 bits.

Another analogy (a gross simplification) that may make things clearer: 
-say you have a computer screen with a resolution of 10,000x10,000 pixels (100 million pixels total, clearly the best computer screen in existence), this screen can also display every shade of colors conceivable (that is actually quite possible with current technology)
-Now you have a digital camera that can take a picture with just as high a resolution and color depth as your monitor (100 megapixel camera, every color imaginable dynamic range)
-you take a picture and save it, then you send it to a friend with the same computer screen that you have, he should be able to see what you see; however you find out that it would take a year to actually TRANSFER the picture; therefore you compress the picture
-there are many ways to compress a picture, however you happen to choose averaging so, you average all 100 million pixels and also you reduce the amount of colors that can be seen to a possible 32 colors (like a crayola box or something)
-after averaging your picture (which happens to be of a green field and a blue sky), you figure out that your picture is an average color of blue-green... this is the information you send to your friend (make your screen blue-green)
-so now your friend sees a screen that is solid blue green... the average of what you both are seeing is the same; however since you have the original, and your friend has the average, all your friend can see is a single colored screen (a big blue green square), where you can see the original picture.
-I would think it is obvious that your friend is never going to be able to *see what you are seeing* with the original image... all he knows is that the average is blue green... if someone came up to me and said 'look at a blue green square', I am not going to know that "oh, they took a picture of a green field and a blue sky"... this is the problem with compression and re-sampling audio, you cannot have better data than you are given.


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