Old 6th March 2014, 14:22   #1
p0rt
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[split] p0rt's audio dicussions...

the equalizer sounds rubbish, it should be a full dsp equalizer, and winamp should have ASIO for its default output in todays world
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Old 6th March 2014, 14:29   #2
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time for you to get coding then...
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Old 7th March 2014, 01:01   #3
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http://www.dsp-worx.de/?n=6 not mine

windows audio works different to windows xp when openAL was the bomb

ASIO bypasses windows and goes straight to your card, all the filters are all differet how they work

the winamp EQ needs some work anyway if the new owners want to bring some sells back to it, in the world of 24bit audio
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Old 7th March 2014, 03:12   #4
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You can get some of those features already....
Go to: Winamp > Prefs > Playback
Playback tab: Checkmark "Allow 24bit"
Equalizer tab: Change EQ Type to "Constant-Q" and Frequency Bands to "ISO Standard"

Much better than the default settings, imo :-)
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Old 7th March 2014, 09:54   #5
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@p0rt
You can get some of those features already....
Go to: Winamp > Prefs > Playback
Playback tab: Checkmark "Allow 24bit"
Equalizer tab: Change EQ Type to "Constant-Q" and Frequency Bands to "ISO Standard"

Much better than the default settings, imo :-)
i know, the equalizer still is`nt that good if you compare it to vst`s etc

and most of the categories on http://www.radionomy.com/en are for audiophilies who would never use winamp if it sounds bad
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Old 7th March 2014, 10:06   #6
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24-bit output / EQ mode doesn't help much (if at all) when most input media is only 16-bit (as padding the upper 8-bits with 0s so adds to the audio quality) and seeing as most streaming audio (as you've used in the above example) has already been re-processed as part of stream generation, applying additional EQ / sample conversion on those is just going to potentially cause more issues.

you seem to be a complete negative bashing of anything and everything to do with Winamp and SHOUTcast currently (is like you're just trying to troll things) and maybe it's not the setup for you to be using.

either way it's known that things are not as wonderful as they could be (and it's going to take time if done to improve things), but there are reasons why things are as they are, yet you cannot magically improve audio quality (as you seem to be trying to say) when the input media is the largest constraint on things. and it's going to take time to attempt to make improvements when we've still got to go through the process of getting Winamp back into a state post-sale which even allows for a new client release to be made.

so in the nicest of ways, cut us some slack for the time being.
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Old 7th March 2014, 10:19   #7
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most audio is 24bit mastered, and what makes the audio file sound twice as loud as 16bit

i have`nt bashed nothing, the way windows audio works is all different to when winamp had its last big update, just using some winamp processor and a dll, will not be as good as DSP`s using your 24bit ASIO soundcard, even if its just the soundcard in a laptop
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Old 7th March 2014, 21:36   #8
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Originally Posted by p0rt View Post
most audio is 24bit mastered, and what makes the audio file sound twice as loud as 16bit
If the file was 24bit mastered, then the bit-depth was changed to 16bits, then there's no benefit of outputting that file to 24bit for listening. It's like converting an mp3 file to FLAC, what's lost is already lost.
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Old 7th March 2014, 21:41   #9
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am so glad someone else posted ^^
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Old 8th March 2014, 06:46   #10
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24-bit output / EQ mode doesn't help much (if at all) when most input media is only 16-bit
If you e.g. "replay gain" contemporary mastered tracks according to the EBU R128 / ATSC A/85 norms, you will observe that they are attenuated typically by about 15-16 dB (EBU R128) / 16-17 dB (ATSC A/85).

If you put into this rough estimation that 6 dB corresponds to a shift by 1 bit it is easy to see that 15-16 dB (EBU R128) / 16-17 dB (ATSC A/85) is close to 3 bit.

The bottom line is as follows: If you attenuate contemporary mastered tracks (hard limited, following the "loudness war", taken from a 16-bit CD) according to EBU R128 / ATSC A/85
  • in 16 bit mode you're loosing up to 3 bits dynamic resolution (there's nothing left where the 3 bits can go to),
  • in 24 bit mode you're loose nothing (because there is a 8 bit "buffer" where the 3 bits can go to).
Whether this is useful or not depends on further circumstances. In case you're DAC is 24-bit ASIO capable, indeed, I suggest using it.
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Old 8th March 2014, 08:53   #11
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Whether this is useful or not depends on further circumstances. In case you're DAC is 24-bit ASIO capable, indeed, I suggest using it.
It is my understanding that enabling Winamp to allow 24-bit output will allow input that is encoded at greater than 16-bit to pass thru while adding enough zeroes (if necessary) to make a 24-bit number. Input encoded as 16-bit will simply have 8 zeroes added to the 16-bit number.

If 24-bit output is disabled then any input encoded at greater than 16-bit is truncated to a 16-bit number.

Are you saying that Winamp's ReplayGain feature will turn a 16-bit number into a 19-bit number and only 5 zeroes will be added when 24-bit output is enabled?

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Old 8th March 2014, 14:40   #12
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If the file was 24bit mastered, then the bit-depth was changed to 16bits, then there's no benefit of outputting that file to 24bit for listening. It's like converting an mp3 file to FLAC, what's lost is already lost.
pro hardware is all 24bit in todays world, audophilie seperates stereo hardware will be all 24bit when DAC have a mini usb standard instead of using RCA leads

radioamy looks like some audiophilie site, and windows audio works differenct to 8 years ago when winamp had its last big update

mp3`s is the poorest audio there is in any bitrate and won over VQF because the filesize was 2mb smaller and easier to transfer over dialup modems

after the top end world gets a standard, all of your tv`s and ipods will become 24bit, and those fail when trying to spot the future before something gets a standard
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Old 9th March 2014, 04:36   #13
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I can't agree with that last statement. 24 bit is overkill in my opinion.
Part of my course was to write a paper and investigate "psycho acoustic" encoding quality for file size / transport and the science behind clarity of the encoded audio.

This and of the blind listening tests I have found on the internet, for tests using the highest quality headphones, 99% of the data collected shows people can't tell the difference when at a high enough bit-rate. The majority being 192+ bits per second in .mp3 "Fraunhoffer encoder"

With regards to huge speaker set-ups like clubs and concerts, subtle harmonic transients are "felt" more in 24 bit non encoded audio but few have a set-up like that.
It is true that an .mp3 file is nothing like the original audio signal after processing however, again, blind listening tests have proven .mp3 can be considered to be perceived as transparent to the source material when encoded at a high enough bit rate, yes, some of the transients are removed and they can't be put back, so what if you can't hear them. there is not a speaker made today that reproduces the sound accurate every time it's played, temperature changes, pressure in the atmosphere, time of day all change the colour of sound being played. That's the whole point of "Psycho acoustic" encoding. the basis of it is centred around what you perceive to hear Vs what you actually hear. The same reasoning around why encode the sound of a pin dropping in a room where a loud drum solo is occurring.

To say the encoded sound is of poor quality is not correct. It sounds transparent to 99% of the people who listen to it at a 10th or more the size taken to store it.
If you are going to compare the difference in actual wave form, the source is completely different to the encoded file and hence, you could say the quality is reduced although, you would never hear that so what's the point?
Are we discussing the quality of how something sounds to humans or, how accurate it is reproduced. You can't hear a dog whistle however, you can record it. What's the point in trying to listen to audio you can't hear.

Either way, AAC and other advances in "Psycho acoustic" modelling audio encoding have tried to address some of the short-falls found in .mp3, not so much to improve the sound, it's has everything to do with making the encoded audio even more transparent at even lower bit-rates than .mp3. AAC is considered to be transparent at 128bits, which is lower than .mp3 at 192. It uses a variable bit rate that juggles space over a single bit per second and not jumps in encoding bit-rates like .mp3 does like 64bits to 128 bits to 192 bits as an example. it can move anywhere in between these values again, making the space required for storage more efficient. There are other advances as well, that's the easiest one to explain.
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Old 9th March 2014, 08:03   #14
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Originally Posted by p0rt View Post
pro hardware is all 24bit in todays world, audophilie seperates stereo hardware will be all 24bit when DAC have a mini usb standard instead of using RCA leads
Again, 24bit hardware means nothing if the source is 16bit.

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Originally Posted by p0rt View Post
radioamy looks like some audiophilie site, and windows audio works differenct to 8 years ago when winamp had its last big update
It looks like it's a regular stream site. Where do you get that it's an audiophile site? Looks like a regular streaming site that's streaming at 16bit to me.
http://board.radionomy.com/viewtopic...t=568&start=20

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mp3`s is the poorest audio there is in any bitrate and won over VQF because the filesize was 2mb smaller and easier to transfer over dialup modems
VQF? I never even heard of VQF ever existed on streaming, and I used to love VQF back in 99/2000 because of the quality versus mp3. VQF had much better quality versus bitrate than mp3, meaning that a 96kbps VQF was as good if not better than 128kbps MP3, at the cost of CPU though, So in comparison, VQF would have been a better candidate for streaming than MP3 in regards to bitrate, but the truth is VQF never caught on and MP3 was already the de facto. There wasn't really many decoders/encoders for VQF anyway.

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after the top end world gets a standard, all of your tv`s and ipods will become 24bit, and those fail when trying to spot the future before something gets a standard
It will be many years before that. We took a step back when ipods became standard because people went from lossless CDs to lossy MP3s. the DACs on the ipods are pretty crappy too. Does the general public even care about 24bit? No. If that was true, SACD and similar would have taken off. The general public doesn't care about being an audiophile, and since the general public determines sales, it will be a very long time till we see 24bit becoming standard for general media consumption.
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Old 9th March 2014, 18:39   #15
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It is my understanding that enabling Winamp to allow 24-bit output will allow input that is encoded at greater than 16-bit to pass thru while adding enough zeroes (if necessary) to make a 24-bit number. Input encoded as 16-bit will simply have 8 zeroes added to the 16-bit number.
Yep.

Quote:
Originally Posted by Aminifu View Post
If 24-bit output is disabled then any input encoded at greater than 16-bit is truncated to a 16-bit number.
Yep.

Quote:
Originally Posted by Aminifu View Post
Are you saying that Winamp's ReplayGain feature will turn a 16-bit number into a 19-bit number and only 5 zeroes will be added when 24-bit output is enabled?
I don't know what WA does.

But after all applying RG is some kind of multiplication, which you can imagine as a bit shift.

In case you're using the FFSoX input plugin, I can assure you that it is exactly as described in my previous post: 16 bit -> 24 bit & RG: no bit is lost.

Quote:
Originally Posted by pbelkner View Post
In case you're using the FFSoX input plugin, I can assure you that it is exactly as described in my previous post: 16 bit -> 24 bit & RG: no bit is lost.
Even a 16 bit -> 24 bit is a multiplication. In FFSoX input plugin 16 bit -> 24 bit & RG is just one 64 bit float operation.
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Old 11th March 2014, 16:27   #16
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Originally Posted by shades_aus View Post
@p0rt
I can't agree with that last statement. 24 bit is overkill in my opinion.
Part of my course was to write a paper and investigate "psycho acoustic" encoding quality for file size / transport and the science behind clarity of the encoded audio.

This and of the blind listening tests I have found on the internet, for tests using the highest quality headphones, 99% of the data collected shows people can't tell the difference when at a high enough bit-rate. The majority being 192+ bits per second in .mp3 "Fraunhoffer encoder"

With regards to huge speaker set-ups like clubs and concerts, subtle harmonic transients are "felt" more in 24 bit non encoded audio but few have a set-up like that.
It is true that an .mp3 file is nothing like the original audio signal after processing however, again, blind listening tests have proven .mp3 can be considered to be perceived as transparent to the source material when encoded at a high enough bit rate, yes, some of the transients are removed and they can't be put back, so what if you can't hear them. there is not a speaker made today that reproduces the sound accurate every time it's played, temperature changes, pressure in the atmosphere, time of day all change the colour of sound being played. That's the whole point of "Psycho acoustic" encoding. the basis of it is centred around what you perceive to hear Vs what you actually hear. The same reasoning around why encode the sound of a pin dropping in a room where a loud drum solo is occurring.

To say the encoded sound is of poor quality is not correct. It sounds transparent to 99% of the people who listen to it at a 10th or more the size taken to store it.
If you are going to compare the difference in actual wave form, the source is completely different to the encoded file and hence, you could say the quality is reduced although, you would never hear that so what's the point?
Are we discussing the quality of how something sounds to humans or, how accurate it is reproduced. You can't hear a dog whistle however, you can record it. What's the point in trying to listen to audio you can't hear.

Either way, AAC and other advances in "Psycho acoustic" modelling audio encoding have tried to address some of the short-falls found in .mp3, not so much to improve the sound, it's has everything to do with making the encoded audio even more transparent at even lower bit-rates than .mp3. AAC is considered to be transparent at 128bits, which is lower than .mp3 at 192. It uses a variable bit rate that juggles space over a single bit per second and not jumps in encoding bit-rates like .mp3 does like 64bits to 128 bits to 192 bits as an example. it can move anywhere in between these values again, making the space required for storage more efficient. There are other advances as well, that's the easiest one to explain.
most people in the world can hear how bad all mp3 bitrates are, 100s and 1000s of clubs have shutdown worldwide in the past 6 years vinyl has`nt been played anywhere and most shops wont sell waves, some that do just transcode your download from a mp3 the label has given them

there is no hope in hell anyone would play a bitrate lower then 320, the hi hz will all sound like a 64bit mp3 and no file with have any sub bass, even 320bit mp3`s sound like pure noise with or without system that ear plugs are recommended for its decibel levels

mp3`s sound listenable on headphones and low budget hifi`s and no where else
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Old 22nd March 2014, 04:17   #17
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Quote:
Originally Posted by p0rt View Post
most people in the world can hear how bad all mp3 bitrates are, 100s and 1000s of clubs have shutdown worldwide in the past 6 years vinyl has`nt been played anywhere and most shops wont sell waves, some that do just transcode your download from a mp3 the label has given them

there is no hope in hell anyone would play a bitrate lower then 320, the hi hz will all sound like a 64bit mp3 and no file with have any sub bass, even 320bit mp3`s sound like pure noise with or without system that ear plugs are recommended for its decibel levels

mp3`s sound listenable on headphones and low budget hifi`s and no where else
I don't believe you at all.
Have you got proof somewhere?
Can you provide reputable sources?
Have you tried theses blind listening tests yourself or with friends?

The tests I am talking about are double blind listening tests at transparency bitrate encoded files.

I have done this myself at University of Technology Swinburne as part of my Electronic Science Diploma.
Mine was not nearly as controlled as others have tested, or as broad a range of subjects however, these tests have not just been done by technological institutes like mine or Fraunhoffer but also by encoding community public like the LAME project and the GPPsycho feely developed psychoacoustic encoding engine developed that was developed because of the cost involved licensing the Fraunhoffer one.

I doubt you could tell the difference, audiophiles can't, 99% of all surveyed couldn't. Actually a broad range of people preferred the .mp3 claiming it to be the source material when they could actually tell a difference. This did not happen until bitrates were reduced and Transparency was compromised.

"Researcher John Meyer has devised an objective index demonstrating that mp3s offer far lower fidelity than either vinyl LPs or standard CDs. And yet this eight-year study at Stanford University shows that prolonged exposure to mp3s leads young listeners to prefer the format. No wonder record producers are despairing."
http://www.metafilter.com/85270/mp3s...lity-the-proof

http://wiki.hydrogenaudio.org/index.php?title=LAME

Can We Really Tell Lossless From MP3?
http://news-beta.slashdot.org/story/...sless-from-mp3

So again, I ask you to clarify what you are discussing here.

Quality of the orginal source recording (mics and technique)
Vs
Encoded sound reconstruction quality analysis when decoded.


OR


Are we discussing HOW IT SOUNDS to humans.
i.e. what sounds better/nicer/preferred to humans

I ask this because the two things are very, very different.

Equipment like Speakers/headphones/live musical playback, Vs recorded with sure mics Vs Senheizer mics Vs Fraunhoffer encoder Vs Lame encoder Vs AAC encoder etc etc are all pointless comparisons when it come to how accurate they are because they ALL contribute to colourisation of sound from the original source.

I think what you will find and I hope you take away from this is, MOST people especially younger people actually prefer the sound of the .mp3 format and it is taking over the world.
AAC format is even better when looking at TRANSPARENCY.

A Live Performance sounds different again however, there is no microphone produced on earth that accurately captures the sound of a Live performance, so again, what's your point?

Last edited by shades_aus; 22nd March 2014 at 06:11.
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Old 25th March 2014, 12:23   #18
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Originally Posted by shades_aus View Post
@p0rt


I don't believe you at all.
Have you got proof somewhere?
Can you provide reputable sources?
Have you tried theses blind listening tests yourself or with friends?

The tests I am talking about are double blind listening tests at transparency bitrate encoded files.

I have done this myself at University of Technology Swinburne as part of my Electronic Science Diploma.
Mine was not nearly as controlled as others have tested, or as broad a range of subjects however, these tests have not just been done by technological institutes like mine or Fraunhoffer but also by encoding community public like the LAME project and the GPPsycho feely developed psychoacoustic encoding engine developed that was developed because of the cost involved licensing the Fraunhoffer one.

I doubt you could tell the difference, audiophiles can't, 99% of all surveyed couldn't. Actually a broad range of people preferred the .mp3 claiming it to be the source material when they could actually tell a difference. This did not happen until bitrates were reduced and Transparency was compromised.

"Researcher John Meyer has devised an objective index demonstrating that mp3s offer far lower fidelity than either vinyl LPs or standard CDs. And yet this eight-year study at Stanford University shows that prolonged exposure to mp3s leads young listeners to prefer the format. No wonder record producers are despairing."
http://www.metafilter.com/85270/mp3s...lity-the-proof

http://wiki.hydrogenaudio.org/index.php?title=LAME

Can We Really Tell Lossless From MP3?
http://news-beta.slashdot.org/story/...sless-from-mp3

So again, I ask you to clarify what you are discussing here.

Quality of the orginal source recording (mics and technique)
Vs
Encoded sound reconstruction quality analysis when decoded.


OR


Are we discussing HOW IT SOUNDS to humans.
i.e. what sounds better/nicer/preferred to humans

I ask this because the two things are very, very different.

Equipment like Speakers/headphones/live musical playback, Vs recorded with sure mics Vs Senheizer mics Vs Fraunhoffer encoder Vs Lame encoder Vs AAC encoder etc etc are all pointless comparisons when it come to how accurate they are because they ALL contribute to colourisation of sound from the original source.

I think what you will find and I hope you take away from this is, MOST people especially younger people actually prefer the sound of the .mp3 format and it is taking over the world.
AAC format is even better when looking at TRANSPARENCY.

A Live Performance sounds different again however, there is no microphone produced on earth that accurately captures the sound of a Live performance, so again, what's your point?
i dont need proof. event and club systems, and rack mounts of compressors and EQ`s all working together, any bitrate mp3 will sound like some trash mono 1940 radio compared to vinyl

mp3 encoding skip most of the needed hz when compressing, so you need the wav most shops dont sell

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Old 25th March 2014, 13:02   #19
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No p0rt, you don't need proof. You know better than anyone else. Your ears are the Gold Standard.

Please start your own thread if you want to continue your lesson. Let this one get back to it's topic.

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Old 25th March 2014, 15:43   #20
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god literally, i reinvent whole scenes, and 99% invented dubstep when dnb died

dubstep was born from a bad EQ`ing 2 am compitition on DNBA production forum
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Old 25th March 2014, 15:48   #21
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when dnb died
never... dnb will never die
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Old 25th March 2014, 15:56   #22
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dnb died in 2003
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Old 25th March 2014, 15:57   #23
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Maybe in your life, but last I checked dnb is alive and well in the scene today
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Old 25th March 2014, 23:34   #24
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it exists, and thats it, its dead in the world of club nights, events, single sells, more then 1000 in the world caring about it

and thats how it will remain forever, or until uk hardcore scene ditches it again from having it in room 2 at their events, while hardcore has whole festivals to itself, and dubstep is dying so trap came along for people who can`t EQ basslines, and trap is dying and hardcore has whole festivals to itself
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Old 26th March 2014, 05:37   #25
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@p0rt
Yes p0rt, you do need proof.
There is a world of people out there and just because you say it is one way does not make it correct.
Inappropriate/bad equalisation / compressing or altering will make a mess out of any audio signal, regardless of the format .mp3 or otherwise. That statement makes no sense in your argument.

There is actually a science behind what I have provided, not just someone's opinion.

I'm afraid I will just have to disagree with you on the topic we discussed, I see no further point in trying to correct your understandings no matter how much actual evidence is going to be brought forward to you.

On a Personal request for improvement to Winamp however, I would love to see an increase in choice of the EQ bands
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Old 26th March 2014, 06:48   #26
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On a Personal request for improvement to Winamp however, I would love to see an increase in choice of the EQ bands
There are already 2 choices. Do you want more choices and/or more bands? Exactly what would you like to have?

I rarely use the Winamp EQ, but I would like more, narrower, bands.

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Old 26th March 2014, 08:11   #27
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finer bands weren't added when the constant-eq mode was added (which would have been the ideal time) due to the limitations implied by the skinning interface (both for skins - especially classic - and the plug-in api). so without breaking a lot of things or providing an alternate ui/api that doesn't fit with anything the current way does (like 3rd party plug-ins have done) I'm not sure that request will ever happen natively with how Winamp desktop currently is.
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Old 26th March 2014, 13:08   #28
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finer bands weren't added when the constant-eq mode was added (which would have been the ideal time) due to the limitations implied by the skinning interface (both for skins - especially classic - and the plug-in api). so without breaking a lot of things or providing an alternate ui/api that doesn't fit with anything the current way does (like 3rd party plug-ins have done) I'm not sure that request will ever happen natively with how Winamp desktop currently is.
How about a new EQ button?
Classic Mode - Normal EQ
HiFi Mode- New EQ with finer/more EQ bands?
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Old 26th March 2014, 13:20   #29
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there's already 2 EQ engines available via the Playback -> Equalizer preferences page which altered the ranges (as already noted in http://forums.winamp.com/showpost.ph...43&postcount=4).

as for a new EQ button. how would that work for the classic skins (fixed format) or all of the modern skins which will never be updated? that is the point i was trying to make in my reply you quoted in that the UI constraints of Winamp from the 2.x days and onwards (especially for classic skin mode which can barely be changed without breaking 1000's of skins).

so it's not something that can be just done without having to seriously take into account the impact it would have on _all_ existing skins (which cannot be updated) and plug-ins which interact with the EQ as-is. and like i said, the only viable way is via a non-integrated window which would be able to do it, but from a usability aspect it would not be a nice experience and just opens up a hornet's nest of pain imho.
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Old 26th March 2014, 19:29   #30
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Originally Posted by p0rt View Post
it exists, and thats it, its dead in the world of club nights, events, single sells, more then 1000 in the world caring about it
Do you really care if a genre of music "exists" in trendy clubs and events? All that matters for music is if someone will make it and someone will play it. Drum & Bass is alive and well in the underground scene here in cali, along with acid, techno, and trance, other genres thought to be "dead".

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Originally Posted by p0rt View Post
and thats how it will remain forever, or until uk hardcore scene ditches it again from having it in room 2 at their events, while hardcore has whole festivals to itself, and dubstep is dying so trap came along for people who can`t EQ basslines, and trap is dying and hardcore has whole festivals to itself
huh? All I took from that was that you are in a different region than me.
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Old 26th March 2014, 19:53   #31
p0rt
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Originally Posted by shades_aus View Post
@p0rt
Yes p0rt, you do need proof.
There is a world of people out there and just because you say it is one way does not make it correct.
Inappropriate/bad equalisation / compressing or altering will make a mess out of any audio signal, regardless of the format .mp3 or otherwise. That statement makes no sense in your argument.

There is actually a science behind what I have provided, not just someone's opinion.

I'm afraid I will just have to disagree with you on the topic we discussed, I see no further point in trying to correct your understandings no matter how much actual evidence is going to be brought forward to you.

On a Personal request for improvement to Winamp however, I would love to see an increase in choice of the EQ bands
there is no scrience if mp3 encoding skips 40% of the hz a single note uses which is how mp3 compresses files

play this http://k007.kiwi6.com/hotlink/tne2a3v3rq/dynamics.mp3

it should sound something like http://k007.kiwi6.com/hotlink/ysjrwwcirb/stylez.mp3 but it will be like http://k007.kiwi6.com/hotlink/eaxeqt...amics_duff.mp3
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Old 27th March 2014, 08:27   #32
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Originally Posted by DrO View Post
so it's not something that can be just done without having to seriously take into account the impact it would have on _all_ existing skins (which cannot be updated) and plug-ins which interact with the EQ as-is. and like i said, the only viable way is via a non-integrated window which would be able to do it, but from a usability aspect it would not be a nice experience and just opens up a hornet's nest of pain imho.
Thanx for explaining why the Winamp EQ is the way it is. I agree that the negative impact to everything else, of adding more bands, should be taken into account.

I only use it for a few of my older mp3s. It would be better for me to find (or make) better encodings of those songs anyway and (like you say) there are plenty of 3rd party plug-ins that offer finer equalization processing (in non-integrated windows).

A lot of people don't understand that EQ and/or DSP processing is not magic. You can't add or enhance anything that is not in the music encoding already. Increasing marginal frequencies too much usually increases the background noise and makes the music sound worse. Decreasing other frequencies to compensate for the marginal ones can kill the dynamics in the music. Equalization should be done lightly or not at all, imo. A poorly encoded file will sound bad no matter what decoding processing is applied to it. All other things being equal, a better sound card/chip and speakers will improve the sound more. Winamp is blamed too much for things it has no control over.

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Old 27th March 2014, 17:43   #33
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DSP`s normally have some kind of processor on your sound card so the EQ is more accurate and faster, more so if your using ASIO which bypasses all of windows sound system and goes direct to your sound card

so then you have to spend £250 for a dedicated DSP card with 3000 processors on
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