View Full Version : Can I transfer Through internet radio broadcast TO FFDSHOW AUDIO ???
vola
30th October 2010, 05:22
Can I transfer Through internet radio broadcast
TO FFDSHOW AUDIO ???
to See if the broadcast can be improved
Snowknight26
30th October 2010, 07:10
Another one of these threads? You can't make 'make' detail. You can't recreate data that doesn't exist. Same applies to audio.
vola
30th October 2010, 08:23
Another one of these threads? You can't make 'make' detail. You can't recreate data that doesn't exist. Same applies to audio.
Another one of these threads?
Something forced you to answer?
You can't recreate data that doesn't exist.
You are wrong the "data " exist
if there not data We can't hear anything
So the question is how you can transfer" radio broadcast"
or "tv broadcast" to ffdshow
Snowknight26
30th October 2010, 18:12
You are wrong the "data " exist
Right, the original audio is data itself. You can't improve it because you don't know how it was supposed to sound originally. Thus, you can't make it sound better. Hence why I quoted
to See if the broadcast can be improved
vola
30th October 2010, 20:45
Right, the original audio is data itself. You can't improve it because you don't know how it was supposed to sound originally. Thus, you can't make it sound better. Hence why I quoted
Yes you can
improve the sound Like that
http://img169.imageshack.us/img169/1283/64560220.png
5.1 Makes an amazing sound
And it also converts the sound to DDL
SO That can be improved
AND the Amplifier Recognize the sound as Dolby Digital
mariush
30th October 2010, 21:24
Your audio source is stereo, 2 channels, of compressed audio.
The person who broadcasts that audio stream encoded it with a software that drops some parts of the audio that your ears can't notice easily to lower the amount of data that comes to you.
As example, if the radio would send the data without any compression to you, just as it is on a CD, it would need 1400kbps - compressed to MP3 or AAC it only needs 96 kbps.
No matter what you do, you won't make it sound "better", like it is on a CD or close to that.
You could make 5.1 sound, but that's not amazing or something that would improve the quality, ffdshow would just copy the signal that would go to front speakers to the ones on the back and then send the bass from both front speakers to the the subwoofer - you'll just hear the poor quality sound on all 5.1 speakers, the radio will not suddenly become 5.1, 3D and so on.
As for Dolby Digital, if you use a mic to record your voice, would it sound better than normal if you'd convert it to Dolby Digital?
No, you'd just convert from a poor quality format to another poor quality format and possible lose even more quality in the process.
Snowknight26
31st October 2010, 05:55
Yes you can
improve the sound Like that
No, you can't, but good luck deluding yourself. The only 'way' you can is if your definition of improve doesn't actually mean what you think it does. You should read about information theory/lossy compression/generation loss and the likes to save everyone a headache.
yetanotherid
31st October 2010, 06:20
Yes you can
improve the sound Like that
http://img169.imageshack.us/img169/1283/64560220.png
5.1 Makes an amazing sound
And it also converts the sound to DDL
Personally I think 5.1 sound sucks and I'd much prefer stereo, but that aside....
Taking ordinary quality 2ch audio and spreading it around over more speakers isn't really a definition of improvement, and it seems you're thinking "more of the same" is an improvement, which is quite odd.
SO That can be improved
AND the Amplifier Recognize the sound as Dolby Digital
Define "improved"?
vola
31st October 2010, 07:47
The person who broadcasts that audio stream encoded it with a software that drops some parts of the audio
Yes I know
So it bothers me and I use - conversion to 5.1
or to 24/96 ( in 24/96 is Increase the bass but the 5.1 ddl
Wonderful with alc 892 gigabyte )
if the radio would send the data without any compression to you, just as it is on a CD, it would need 1400kbps - compressed to MP3 or AAC it only needs 96 kbps.
I know so
and the radio send me 64 kbps Usually
you write me Theories that i know
No matter what you do, you won't make it sound "better", like it is on a CD or close to that.
Not like in CD
But it gets better than " radio broadcast" or "tv broadcast"
and if i Could move to FFDSHOW This will improve further
"mariush" You must learn to think outside the box
Theories Paralyzing you
You could make 5.1 sound, but that's not amazing or something that would improve the quality
Big mistake
5.1 sound great, you do not have my hardware, did not hear my
sound so Theories again ?
ffdshow would just copy the signal that would go to front speakers
You do not know how I analyze the sound with FFDSHOW
You write off the point, and not true !
As for Dolby Digital, if you use a mic to record your voice
Again you are making and write off the point
No, you'd just convert from a poor quality format to another poor quality format and possible lose even more quality in the process.
Some nonsense? Better quality improves not perfect But better
No, you can't, but good luck deluding yourself.
not need luck
The only 'way' you can is if your definition of improve doesn't actually mean what you think it does
You are able to understand what you wrote? You know what i think it does ?
You should read about information theory/lossy compression/generation loss
I also understood them, I'm afraid you do not understand
Define "improved"?
As a person who does not like 5.1 You're the only one Asks to the Point!
I had many computers Only with ALC 892 I felt improvement in Surround sound ( ONLY GIGABYET ASUS Sound is not good with the same chip they can't make 5.1 )
Distribution to 7.1 is seamless Orgyanal
ALC 892 first Chip just doing a wonderful job.
Here there is no improvement in bass , Improvement
in Every component in the sound this wow !!!
Today I'm taking 5.1 DD sound OR DTS 5.1 1500 KB(dts Too much bass for me )
Puts to FFDSHOW and Make changes and converts to "resample to 192000hz " in streo
> from there to Sound Card Converted to 5.1
Believe me better than the original !
leeperry
31st October 2010, 08:12
upsampling is a mixed bag, some strongly believe in it: http://photos.imageevent.com/cics/v03theartofbuildingcomputertrnsp/The%20art%20of%20building%20Computer%20Transports%20v0.3.pdf
http://thumbnails24.imagebam.com/10461/19df71104603551.jpg (http://www.imagebam.com/image/19df71104603551)
this said, 64kbit mp3 off a realtek via DirectSound will kill most -if not all- of the aforementioned supposed improvement IMHO.
You didn't say what player you use to listen to those web radios...hard to say. You can force ffdshow audio in a DirectShow player rather easily.
GodofaGap
31st October 2010, 10:11
http://thumbnails24.imagebam.com/10461/19df71104603551.jpg (http://www.imagebam.com/image/19df71104603551)
The only thing you can show with pictures like that is that you don't know how a DAC works.
(HINT: It doesn't connect the dots with straight lines)
leeperry
31st October 2010, 15:31
The only thing you can show with pictures like that is that you don't know how a DAC works.
(HINT: It doesn't connect the dots with straight lines)
well, a proper DAC chip has an oversampling stage that will do just what this picture shows.
This PDF is from the cMP author, and some ppl claim that its SINC upsampling sounds great...I guess it boils down to the DAC chip you're using, its built-in post-filtering and so on. There's many VST plugins that oversample internally to improve their accuracy in the trebles. I already quoted all those in that post: http://forum.doom9.org/showpost.php?p=1452843&postcount=106
Some ppl believe that upsampling improves the DAC post-filtering accuracy, and some that it will only feed bogus extrapolated data. If there's one thing I've learned in audio, it's that there's no hard rule as to what sounds "good". OTOH, 64kbit and Realtek turn it into a risky business :p
yetanotherid
31st October 2010, 16:17
5.1 sound great, you do not have my hardware, did not hear my sound so Theories again ?
What exactly is so special about 5.1?
As movie audio I can understand it, although personally I've never seen the logic behind having the sound coming at you from all directions to keep reminding you the picture doesn't, but when it comes to listening to music how does it make it better? Especially if the music was stereo originally.
As a person who does not like 5.1 You're the only one Asks to the Point!
I didn't really have the question answered though. Admittedly I found your reply a bit hard to follow, but while I understand you're saying it might sound better, you still haven't really said in what way. Clearer? More fidelity?
vola
1st November 2010, 07:33
Thanks for the article
upsampling is a mixed bag,
if this mixed bag Do only bugs :)
Old article from 2007 is not relevant today hardware .
from Your article:
Human hearing is impressive with
an incredible ~132db dynamic range and sensitivity between 20-20kHz!
Studies have that People can hear over 20khz
Your article takes 46 pages you want to say?
The article does not talk about my method so?
this said, 64kbit mp3 off a realtek via DirectSound will kill most
the article Do not write on realtek
There are different processors
Most of which does not support multimedia processing in 2007.
like 3dnow OR 3DNOW+ and AMD cpu .
Intel processors was terrible for multimedia
so the tested was on what cpu ?
http://img201.imageshack.us/img201/5476/75105139.png
this not cpu for multimedia !
So this article Not suitable for today
the Processors are supposed to analyze the information of sound
All your article does not speak or know the difference between processors
Difference between the processor contains processing languages
in 2007 Was processed with only AC'97
Today there is a new Alagoritm HD that can run in BIOS on your cpu ......
http://en.wikipedia.org/wiki/AC%2797
This article talks about WIN XP today have a new system that supports a different sound And better
Sound cards today are better than 2007
This article needs to disqualify with one reason I gave you more than one reason
You didn't say what player you use to listen to those web radios.
Player of the site, not wmp
What exactly is so special about 5.1?
in alc 892
Quality, running the better surround 7.1
More accurate, sharper, better and more fun .
but when it comes to listening to music how does it make it better?
Depends on what music and what comes the Source .
to me Today Halagoritm make better music sound to 7.1 speakers
Do not forget that converts to DDL gives a kind of magic to sound
You have to remembe
1. Intel High Definition Audio Contributes to quality sound ! and to upsampling
2. DDL = Dolby Digital Live When Analysis leaves little mistakes .( Maybe an electric device Can see but not heard in the ear )
http://en.wikipedia.org/wiki/Dolby_Digital
mariush
1st November 2010, 09:13
Studies have that People can hear over 20khz
Yes, young humans can hear from about 5 Hz to about 24 kHz. As people get older, starting from about 30-40 years, they start to lose the ability to hear frequencies that high, going down to about 18-19.5 khz. Old people can barely hear up to 16 kHz.
As a fun fact, that's why the Audio CD inventors chose 44100 Hz for the format, because it's twice the 22050 Hz frequency, considered the maximum the majority of people would be able to hear.
See Niquist's Theorem for more technical stuff : http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem
When broadcasted over the Internet, the audio is sent as a highly compressed AAC or MP3 file. In order to maintain quality at such low bitrates, encoders remove or flatten the high frequency area of a sound - even a 128kbps mp3 is unlikely to contain sound above 18 kHz.
Below 96-128kbps, encoders would also cut down the low frequencies.
A few posts above you said your radio station is 64 kbps - I doubt that audio track would have sound below 50-60Hz and above 16 khz.
Also, poor quality sound cards and onboard sound cards also cut the low frequencies - usually you won't hear anything below 10 Hz. Even your Realtek chip has this issue - the chip may be good but it's connected on the motherboard with wires that go some distance to the jacks on the side of the motherboard, and these are under electrostatic influence from the rest of devices so manufacturers have to add very small capacitors at the output of the jacks to prevent occasional crackling and popping.
So conclusion is while yes, humans can hear up to 20kHz, you're talking about a music/audio track that's between 40 Hz - 18 Khz at best. Playing it on a card capable of 20kHz won't make it magically re-create the audio data that the encoder removed before compressing to MP3.
the article Do not write on realtek
There are different processors
Most of which does not support multimedia processing in 2007.
like 3dnow OR 3DNOW+ and AMD cpu .
Intel processors was terrible for multimedia
so the tested was on what cpu ?
http://img201.imageshack.us/img201/5476/75105139.png
this not cpu for multimedia !
So this article Not suitable for today
the Processors are supposed to analyze the information of sound
The MP3 format is a very simple format to decode - most portable mp3 players out there on the market probably have the processing power of a 166 Mhz pentium mmx.
Any processor, no matter what special instructions they have, will decode the MP3 file the same. So if you open a MP3 file on a 486 and save it to WAV, and then open the same on a six core 1000$ Intel, the files would most likely be IDENTICAL.
The article doesn't say anything about Realtek because Realtek is the sound card. The sound card is not involved in decoding the MP3 file - it just receives the already decoded audio file from the software and it sends the audio signal to the speakers.
3d Now and other processor instructions are not really doing anything special
All your article does not speak or know the difference between processors
Difference between the processor contains processing languages
3d now, as the name says, was designed for 3D stuff - the graphical 3D, not the audio.
The article doesn't mention differences between processors, because they're not important - the software (that takes the MP3 file and converts it to audio to send to the sound card) will detect what additional instructions each processor supports and use them, otherwise it will fall back to another way of achieving the same result.
For example, if you have a Pentium 3, the MP3 decoder may detect and use SSE extensions to decode the MP3 file, if you have a Pentium 2 it may only use MMX. The audio it outputs will be (almost) identical, the only difference is that the decoder will use less processor time when using SSE.
in 2007 Was processed with only AC'97
Today there is a new Alagoritm HD that can run in BIOS on your cpu ......
http://en.wikipedia.org/wiki/AC%2797
Nobody processed anything with AC'97, you're making a confusion.
AC'97 is simply a standard made up telling companies that if they make a chip that can output at least 5.1 audio channels, have a mic jack and a line in jack and respond to the computer in a specific way, then other chips and components in the computer will recognize it as an audio soundcard.
You still needed a driver, which would sit between a software player and the sound card and take the decoded audio from the decoder and send it to the soundcard to play it on speakers.
There is no algorithm in BIOS - in fact BIOS will be gone completely in a few years.
This article talks about WIN XP today have a new system that supports a different sound And better
Sound cards today are better than 2007
This article needs to disqualify with one reason I gave you more than one reason
It's not different sound and better, just another way of passing the audio data between software and sound card. Each method had its good things and bad things but the new one isn't really any better than the old ones - depends on your needs.
in alc 892
Quality, running the better surround 7.1
More accurate, sharper, better and more fun .
It takes crap and generates surround 7.1 crap.
The sound card receives stereo sound from the MP3 decoder - the driver tells the sound card to take that second of stereo sound and convert it to 7.1 using some algorithm built inside the sound card.
If you give the sound card the 64kbps we talked about that's between 50 Hz and 16 kHz and stereo, what do you think it will play on that .1, the subwoofer speaker? It can't generate stuff that's not there.
The surround just gives you a feeling it sounds better because there's more volume and noise reaching your ears from around you - but it's not better quality by any means.
Depends on what music and what comes the Source .
to me Today Halagoritm make better music sound to 7.1 speakers
well yes, quality music will sound better with more speakers, compared to only 2, but it's nothing related to surround or algorithms. Get an amplifier and 20 speakers and you'll also think it sounds better than 7.1.
leeperry
1st November 2010, 13:12
Player of the site, not wmp
you need to use Virtual Audio Cable, set it as the main windows soundcard, then use graphedit and plug VAC > ffdshow audio > Reclock(in KS/WASAPI) :)
SQ in 64kbit MP3 is so lousy that I'd advice to "enhance" it using OzoneMP in ffdshow....you're looking for magic, it can do that if you enable its tube amplifier.
yetanotherid
1st November 2010, 14:00
Studies have that People can hear over 20khz
Which ones? Probably the same studies which have show monster speaker cable worth $27 per inch will gold plated speaker connectors sounds better than ordinary speaker cable.
By the time you're an adult, most people can't hear anything above about 16khz, and even if they could you can be pretty sure your speakers aren't producing it.
There are different processors
Most of which does not support multimedia processing in 2007.
It sounds like you've decided different processors produce different sound for some reason, then eventually imagined you can hear a difference which doesn't exist.
Sound cards today are better than 2007
Yeah, so doesn't that mean the quality of your stereo sound has increased too, or has stereo quality stagnated while surround sound has managed to increase in fidelity?
Quality, running the better surround 7.1
More accurate, sharper, better and more fun .
I remember when people used to use the "environment" settings on their all-in-one hi-fi gear years ago. You know, when you could select the "concert hall" or "small room" settings etc. Many people used them because they were sure it improved the sound, these days it's stereo to 5.1 conversion instead.
Do not forget that converts to DDL gives a kind of magic to sound
No it doesn't. Not even close. A really nice transparent stereo image is where the magic's at, not some pseudo surround annoyance inflicted on a stereo track.
ramicio
1st November 2010, 14:58
64 kbps anything is a joke, even for just speech. I don't give a crap about all you people's math and theorems. Math only applies in a perfect world. You bash anyone who like anything more than CD audio. Go back to HA, commies. Not everyone in the world has ear wax packed ears and can't hear above a certain point. I just love it :angry: when people bring up top end frequency response and the human ear when anyone mentions anything higher than 16/44.1 without considering resolution, and then they whip out a god damn wiki to their precious Nyquist-Shannon theorem. A theorem is not a mathematical rule! It's a statement. It is not definite like 1+1=2, especially when things have errors.
Vola, you need to be less delusional, or at least learn proper English so maybe you will get what people are saying to you. If you have 64 kbps audio from the radio you are never going to make it sound good. You can play it over your 5.1 system, but there is no improvement. More speakers doesn't mean better sound. It is just getting more processed. You can't make sound be better, unless you invented some kind of magical hardware that can know like a God what the original signal was. Take a bitmap and upscale it, and you can fill in pixels to smooth things out and make it not look upscaled, but that takes intelligence, things computers don't have.
nurbs
1st November 2010, 16:27
A theorem is not a mathematical rule! It's a statement. It is not definite like 1+1=2, especially when things have errors.
So, what's your opinion on the Pythagorean theorem (http://en.wikipedia.org/wiki/Pythagorean_theorem)? Is that definite or just something only HA commies would accept as being valid, like Nyquist-Shannon?
In maths theorems are proven statements.
ramicio
1st November 2010, 16:47
In mathematics, a theorem is a statement which has been proven on the basis of previously established statements, such as other theorems, and previously accepted statements, such as axioms. Sounds like circular logic to me. Douche. Not proven physically. Try the pythagorean theorem on something physical and there will ALWAYS be slight rounding. At the atomic scale make a triangle with a 3, 4, and 5 side (lowest number integers to work with the theorem) and I guarantee you the right angle side will not be truly right. It may not be measurable but the real world is not perfect. Math is perfect on paper, but never in the physical sense. Your precious math can bolster your theories in your own little make believe world all you want it to but more people live in the real world and realize if something is captured in a higher resolution (audio or video) the better it will be. Live in your own little world and keep supporting the CD movement. Let those albums keep getting louder. Stick to your flawed ABX tests. I know what I see and hear.
ABX tests are more subject to psychological effects than others that claim anything above lossy "transparency" is psychological. You get tired of hearing the same god damn sample over and over, so you just start out doing well, and then your brain gets sick of redundancy and just starts giving no effort to comparisons and starts guessing. A pretty good tool for assholes with a purist mathematical agenda.
nurbs
1st November 2010, 21:33
If you can't measure a difference you can't prove that there is a difference at all.
Your distinction between a theorem on paper and in the physical world doesn't come into play with the Shannon-Nyquist theorem in digital audio anyway. Digital audio happens on paper. If you have a 48 kHz PCM file it will be able to store frequencies up to 24 kHz no matter how much you object to it (http://www.hydrogenaudio.org/forums/index.php?showtopic=84208&view=findpost&p=726646).
What you call circular logic, what you think doesn't work in the real world forms the basis of digital audio processing. It is what was used to develop the hardware, software and file formats that are currently in use.
Your objection to ABX tests is also invalid. There is no requirement to complete the test of a file in one sitting. You can just as well wait a day or a week between each trial.
By the way I agree with you that capturing at high resolution can't hurt. Where I don't agree with you is that storing/recording at a higher resolution than the source helps quality. To quote you
You can't make sound be better, unless you invented some kind of magical hardware that can know like a God what the original signal was.
That's why
a) (http://forum.doom9.org/showpost.php?p=1426051&postcount=31) storing vinyl (~12bit/38kHz) as 24bit/192kHz and
b) (http://forum.doom9.org/showpost.php?p=1451943&postcount=81) upsampling your 48 kHz audio to 96 kHz (with a 192 kHz intermediate file)
,both things you do, make absolutely no sense at all.
ramicio
1st November 2010, 21:48
And I say again and again if you think you can record music digitally and have it converted back to analog PERFECTLY then you are sadly mistaken. I don't care what all your math tells you. It's like to you music is made of only sine waves and there can be no other shape of waves. Your along with everyones hate of the vinyl format and these arbitrary measurements disgusts me. Noise, wear, and lower frequency response are something that was measured many years ago. You really think there are any turntables from the 60s and 70s that are as good as the best ones now? No. Technology advances all around, but OH NO not for vinyl...hahahahah. Vinyl is on the rise again and for a good reason. It sounds better without spending as much money. Tracking force is so light these days that records actually might last a lifetime. I already stated that I just don't upsample to expect better quality from one thing, I do it to mix audio. You make no sense at all and you are the kind of sissy that gets the bare minimum of what is decent. I like the best, and possible the overboard. I will say again if the DVD medium would have been around they would have chosen a higher bit rate, maybe 24/176,400. They chose 16/44,100 because it fit a decent amount on CD and it was the best frequency that matched NTSC or PAL video recorders. We are in the age of computers making the music now, so any format really could be chosen. I want something better than CD ultimately. Never did I ever say I store vinyl at 24/192. I record and save 24/96. You can argue all you want about the theoretical quality of vinyl, but it's a better spending of money than CD. Too bad you can't download vinyl like you can a CD.
nurbs
1st November 2010, 23:29
And I say again and again if you think you can record music digitally and have it converted back to analog PERFECTLY then you are sadly mistaken. I don't care what all your math tells you. It's like to you music is made of only sine waves and there can be no other shape of waves.
I don't. I think that once the audio has been converted to digital it can be stored perfectly and I think the current technology is good enough so that the conversion to digital or back to analog doesn't introduce any audible artifacts if done properly. Your post here and on Hydrogenaudio have made it pretty clear that you neither know nor care much about the mathematical background of audio processing and yes, mathematically music is a superposition of waves.
Your along with everyones hate of the vinyl format and these arbitrary measurements disgusts me. Noise, wear, and lower frequency response are something that was measured many years ago. You really think there are any turntables from the 60s and 70s that are as good as the best ones now? No. Technology advances all around, but OH NO not for vinyl...hahahahah.
I don't hate vinyl, I have an average sized collection myself, but I like to keep myself informed about the limitations of the formats I use. Vinyl has definitely improved since the 60s, mainly in frequency range, but those improvements weren't dramatic, unless you are talking about alternative vinyl formats like CD-4, but those aren't around anymore. There are also drawbacks to this since especially the higher frequency suffer greatly from wear and dust. The numbers I gave are what you can expect from the average vinyl record today. At the end of the day grooves only have a certain accuracy and a stylus can only move so much before jumping out of the groove. That's where those limitations come from.
Vinyl is on the rise again and for a good reason. It sounds better without spending as much money.
The cost differences on the low end are minimal. If you want a good record player you are going to pay more than for a good CD player. Stylii wear out over time too and good ones aren't that cheap. There isn't a big difference regarding the cost of the records themselves either.
I just don't upsample to expect better quality from one thing, I do it to mix audio.
Yes, but
a) you still keep a track that has a higher sample rate than your source and
b) several people in the thread have explained to you why upsampling won't offer any benefit for the mixing.
I want something better than CD ultimately.
In that case vinyl isn't what you want. Anything that can recorded on vinyl (again the format that is in use today, not failed new ones like CD-4) can be recorded on CD in higher fidelity while at the same time avoiding the noise and wear the playback of vinyl suffers from. The problem today lies with the mastering.
Too bad you can't download vinyl like you can a CD.
Well yes and no. You can record it digitally and then download that :D. The drawback is that the noise will not change every time you play it. The benefit is that the record will not degrade slightly every time you play it.
GodofaGap
1st November 2010, 23:29
And I say again and again if you think you can record music digitally and have it converted back to analog PERFECTLY then you are sadly mistaken.
Surprise surprise, you can't do this with analog either.
Blue_MiSfit
1st November 2010, 23:35
Gentlemen, please keep the discussion civil and on-topic.
mariush
2nd November 2010, 00:05
I'll second what nurbs said about the vinyl... and I make some additions.
There are vinyl players out there that use laser to read the disk grooves instead of needles or whatever they're called - they're very expensive yet very appreciated and if I remember correctly they were used by some library to create backups of their discs : http://www.vinyldisc.co.uk/ELP.htm
Vinyl discs don't come back in fashion because of the higher quality, they shouldn't be of higher quality compared to a CD but because of the loudness war (http://en.wikipedia.org/wiki/Loudness_war) some CDs really do sound worse than vinyls when played on good audio systems. With vinyl it's just harder to do that loudness thing compared to CD so they "feel" better when listened.
Some say the sound of a vinyl is also "warmer", because of the analogue amplifier but that's debatable.
IMHO they're coming back in style more because some bands want to be "cool" or because they bundle one of these harder to pirate vinyls with regular CDs, so that they sell more physical copies.
I doubt they think about quality, but more like "harder for a drug addict to steal a vinyl off the shelves and stuff it in his jeans"
Anyways, quality wise, no matter how hard you try, a vinyl will eventually get dirty, get easier to break or bend due to air humidity (or lack of) and other factors - makes more sense to digitize it at a high sample rate and preserve them digitally.
Something better than CD it is not... some artists like Mike Oldfield (http://www.amazon.co.uk/Tubular-Bells-2003-DVD-AUDIO/dp/B0000AGPXM) produced DVD Audio or Bluray discs with their tracks but they actually re-mastered the tracks from original material, to take advantage of the medium... unfortunately there's few artists out there (I'm thinking Enya maybe or mostly opera music ) that would benefit a good deal from the better format.
vola
2nd November 2010, 07:51
Yes, young humans can hear from about 5 Hz to about 24 kHz
I have an ear examination that I hear over 20KHZ
I'm not 5 years old
poor quality sound cards and onboard sound cards also cut the low frequencies
Intel High Definition Audio- He opened and analyzes the sound quality and whan you work with (Intel High Definition Audio)
irrelevant sound card that work wite ( ac97 that analyzes in the card) The big difference is here
to Hear low frequencies you need Open the option in windows, this
Just simple Windows default is 80 HZ ( try 10 hz)
( First make sure the speakers or amplifier can work with low frequencies )
the Processors are supposed to analyze the information of sound
Everything has better and worse
in 2007 the cpu in the Article worse !!!
Any processor, no matter what special instructions they have, will decode the MP3
Again No Intel High Definition Audio is
http://en.wikipedia.org/wiki/Intel_High_Definition_Audio
Command or program uses the processor's languages and give you Better results !
( like you say Fiat's is a car and Mercedes is a Car and Both Equal in quality and they Are not ) .
3d Now and other processor instructions are not really doing anything special
wow Are you kidding?
3d now, as the name says, was designed for 3D stuff
wow Are you kidding?
My name is Master Big so i have Big......
http://img829.imageshack.us/img829/535/69784584.png
why FFDSHOW have the language 3DNOW +
For three dimensional ? no for video and audio ( for now ).
you need to use Virtual Audio Cable, set it as the main windows soundcard, then use graphedit and plug VAC > ffdshow audio > Reclock(in KS/WASAPI)
I did not understand anything ?
this is Internet radio not Instrument radio that I put in the computer Through cable you Understand ?
It sounds like you've decided different processors produce different sound for some reason, then eventually imagined you can hear a difference which doesn't exist.
The different languages on CPU they analyze a different way the same sound
So it sounds better or less good
in 2007 intel had "Processor's Nightmare " for Sound .
the "Processor in the Article "Processor's Nightmare " !!!
I can not hear something that does not exist
I can write software to analyze something that exists .
that Old software can not analyze all range of sound and
Not too good quality
Yeah, so doesn't that mean the quality of your stereo sound has increased too, or has stereo quality stagnated while surround sound has managed to increase in fidelity?
that mean Analysis of better sound more accurate
Surround analysis works with a new algorithm is more accurate than in the Past
I remember when people used to use the "environment" settings on their all-in-one hi-fi gear years ago. You know, when you could select the "concert hall" or "small room" settings etc. Many people used them because they were sure it improved the sound, these days it's stereo to 5.1 conversion instead.
in the Past !
Today a newer method
If you have AC 97 You may be need the "environment" in the Computer .
GodofaGap
2nd November 2010, 08:06
Vinyl discs don't come back in fashion because of the higher quality, they shouldn't be of higher quality compared to a CD but because of the loudness war (http://en.wikipedia.org/wiki/Loudness_war) some CDs really do sound worse than vinyls when played on good audio systems. With vinyl it's just harder to do that loudness thing compared to CD so they "feel" better when listened.
Some say the sound of a vinyl is also "warmer", because of the analogue amplifier but that's debatable.
For me, the only major attraction to vinyl is the larger artwork.
Getting better sound is nowadays is not a problem of bitdepth or sampling rate anymore. 44.1/16 or perhaps 24 for the really anal is more than enough. The real gain is to be had in real multi-channel audio (not silly upconverted stuff), but seemingly only a small portion of the populace cares when its not a movie.
mariush
2nd November 2010, 08:26
It feels like I'm talking with someone who mixes everything in his mind.
I have an ear examination that I hear over 20KHZ
I'm not 5 years old
If you hear over 20 kHz, then good for you - you're a superhuman. I'm sure you also have speakers capable of producing that frequency - they're called tweeters and they look like this : cheap speakers don't have such speakers.
poor quality sound cards and onboard sound cards also cut the low frequencies
Intel High Definition Audio- He opened and analyzes the sound quality and whan you work with (Intel High Definition Audio)
irrelevant sound card that work wite ( ac97 that analyzes in the card)
You're mixing up or confusing things again.
AC 97 and Intel HD Audio are specifications - they're not processors, but words on a paper. Just like saying something is a car when it has at least 3 tires, one engine and one steering wheel, Intel HD Audio and AC'97 say some chip is an audio soundcard if that chip receives input and outputs something to speakers.
The difference between AC'97 and Intel HD Audio is simply that the later says a chip must be powerful enough to be able to play sound that's 32 bit, 192 kHz. AC '97 chips don't have to support sound at that parameters but they may be able to, or may be able to support just 24 bit, 192 kHz which is enough anyway. So an AC'97 chip can also be Intel HD audio chip.
Again, please understand, the chip, the small processor that receives the sound from computer programs and sends it to the speakers, can be made by various companies and they're all of various qualities. Those things you talk about are just some rules that say that the chips must be capable of a few things. In your case, it's Realtek ACL882 or something like that, but just like this chip there's also Via Envy, Creative EMUsomething, all are Intel HD Audio or AC'97 and have various degrees of quality.
The quality of the sound that comes out of a sound card is influenced a lot by the electrostatic charges and overall "noise" inside the computer - all the chips inside the computer generate radio frequency and other "bad" frequencies that can affect the quality of the sound. Soundcards that are on a motherboard have to sit very close to other components like video cards so companies know they'll be influenced and on the wire that goes from the chip to the jack connector in the back this "bad stuff" adds up - the wire acts like a radio antenna and the stuff it receives is like a very low humm so manufacturers add electronic components called capacitors just before the speaker jacks. These components make it so that anything below a certain frequency can no longer be heard, so even if you set in the sound card's software to hear as low as 5 Hz, you won't hear anything because those capacitors can't be turned off and will cut everything below a value.
Separate sound cards can produce better quality and are less affected by this because they usually have shielding and have gold connectors. like this one : http://www.newegg.com/Product/ImageGallery.aspx?CurImage=29-132-014-Z03&SpinSet=29-132-014-RS&ISList=29-132-014-Z01%2c29-132-014-Z02%2c29-132-014-Z03%2c29-132-014-Z04%2c29-132-014-Z05%2c29-132-014-Z06&S7ImageFlag=1&Item=N82E16829132014&Depa=0&WaterMark=1&Description=ASUS%20Xonar%20Essence%20ST%20PCI%20Interface%20Audio%20Card
http://img829.imageshack.us/img829/535/69784584.png
why FFDSHOW have the language 3DNOW +
For three dimensional ? no for video and audio ( for now ).
ffdshow is a CODEC that decodes VIDEO and AUDIO. 3D Now! are graphical instructions so they may be used by ffdshow to decode various video formats. Just because it's checked there, it doesn't mean it's used by everything including audio. ffdshow also decodes subtitles, series of pictures - what would 3d now be of use there?
For MP3, ffdshow probably uses only MMX, SSE and SSE2. There's no need for the rest to decode MP3 files.
If you look at http://en.wikipedia.org/wiki/3DNow! , you will see the majority of the 3D Now instructions are in regard to working with floating point numbers - like 3.15879 for example - they're not useful for decoding MP3 files which use very little floating point. Also, 3D Now is deprecated, meaning it's forbidden to be used by new programs and will not work on 64 bit processors. So ffdshow will probably not use it at all if you have a 64 bit operating system.
nurbs
2nd November 2010, 11:51
@mariush:
Do you have any experience with that card you mentioned or Asus in general? I'm in the market for a new soundcard and I wonder how their driver support is.
leeperry
2nd November 2010, 12:21
Asus in general? I'm in the market for a new soundcard and I wonder how their driver support is.
http://brainbit.wordpress.com/2010/07/19/asus-xonar-unified-drivers/
I don’t know what those monkey coders at Asus and C-Media are doing.
TinTime
2nd November 2010, 12:35
@mariush:
Do you have any experience with that card you mentioned or Asus in general? I'm in the market for a new soundcard and I wonder how their driver support is.
I've got one of these (http://uk.asus.com/product.aspx?P_ID=QsEKBPr6ko9pFF2D&templete=2). The hardware's excellent but the drivers are dreadful. I mean just awful. Leeperry's monkey quote is spot on.
nurbs
2nd November 2010, 13:11
Yes, the problem is usually the drivers, not the hardware. I think I'm going to make a thread in the PC Hardware subforum here before I decide. Thanks.
yetanotherid
2nd November 2010, 23:38
I have an ear examination that I hear over 20KHZ
I'm not 5 years old
Well those two claims are certainly contradictory.
And even if you could, human hearing isn't flat. Even if you could hear 20khz the response to frequencies declines as they increase so by the time you hit 20khz you'd be virtually deaf.
Once again, what speakers are you using? My bet is they don't have a flat response up to 20khz even if they can produce frequencies that high.
Intel High Definition Audio- He opened and analyzes the sound quality and whan you work with (Intel High Definition Audio)
irrelevant sound card that work wite ( ac97 that analyzes in the card).....
Again No Intel High Definition Audio is
http://en.wikipedia.org/wiki/Intel_High_Definition_Audio
Command or program uses the processor's languages and give you Better results !
You've no idea what you're talking about.
The different languages on CPU they analyze a different way the same sound
So it sounds better or less good
You've no idea what you're talking about.
Surround analysis works with a new algorithm is more accurate than in the Past
More accurate compared to what? The original stereo sound you're wanting to convert to surround sound? How do you accurately convert stereo to surround?
in the Past !
Today a newer method
If you have AC 97 You may be need the "environment" in the Computer .
That was my point exactly. People can fool themselves into thinking something sounds better just because they think it should. Using the "environment" settings on a computer is another example.
Anyway, no point continuing the debate for me because I don't have super-human hearing like you. I can't hear up to 20khz or hear the difference in the sound of music encoded and decoded using different CPUs.
vola
3rd November 2010, 06:13
AC 97 and Intel HD Audio are specifications - they're not processors,
wow where i say They processors ?
The difference between AC'97 and Intel HD Audio is simply that the later says a chip must be powerful enough to be able to play sound that's 32 bit, 192
NO
In your case, it's Realtek ACL882
NO
I HAVE 892 NOT 882
mariush
I'm sorry you write so many things wrong
You take things out of context
I do not have the time to explain every detail to you
The question was:
Can I transfer Through internet radio broadcast TO FFDSHOW AUDIO ???
ffdshow is a CODEC that decodes VIDEO and AUDIO. 3D Now! are graphical instructions so they may be used by ffdshow to decode various video formats. Just because it's checked there, it doesn't mean it's used by everything including audio
NO
For MP3, ffdshow probably uses only MMX, SSE and SSE2
NO
" probably" NO
they're not useful for decoding MP3 files
NO
You've no idea what you're talking about.
yetanotherid That's what I think of you
More accurate compared to what? The original stereo sound you're wanting to convert to surround sound? How do you accurately convert stereo to surround?
You write You've no idea what you're talking about
SO If I do not know why you ask me?
If you have AC 97 You may be need the "environment" in the Computer .
The intention was to " NO NEED " I've forgotten the NO
I don't have super-human hearing like you
It's true
Ghitulescu
3rd November 2010, 09:53
If you can't measure a difference you can't prove that there is a difference at all.
Your objection to ABX tests is also invalid. There is no requirement to complete the test of a file in one sitting. You can just as well wait a day or a week between each trial.
That's why
a) (http://forum.doom9.org/showpost.php?p=1426051&postcount=31) storing vinyl (~12bit/38kHz) as 24bit/192kHz and
b) (http://forum.doom9.org/showpost.php?p=1451943&postcount=81) upsampling your 48 kHz audio to 96 kHz (with a 192 kHz intermediate file)
,both things you do, make absolutely no sense at all.
Agree. This is the technicians' view.
Disagree. It's so easy to fake an ABX test.
Agree. Overkill for a, nonsense for b.
nurbs
3rd November 2010, 11:38
I agree that ABX tests are easy to fake, it's trivial if they are unsupervised, but I was only saying that that one claim he made was wrong.
If you do it to test your own setup you have no reason to fake it, because you are only deluding yourself.
If you want to convince other people you can always post samples so they can repeat the test.
In the end it's important to know the limitations of the method.
An ABX test is a valid tool to find out if an individual can reliably tell the difference between the two tested files on the hardware used for testing, if done properly. Nothing more, nothing less.
If you have very good hardware and test large numbers of people you get an idea if it's generally possible to do stuff (24bit -> 16bit; 96kHz -> 48kHz; using a more efficient psy model; ...) without the majority of testers noticing.
Ghitulescu
3rd November 2010, 11:46
Yes, if one does the ABX test for himself/herself, as you said, s/he has no intention to fake it.
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