BeSweet to support SSRC ???

[ChristianHJW]

... i just found this mentioned in the ultra mega thread about BeeSweet ( AC3ToMP3 formerly ), but i really dont want to pump up this thread even more .... i found nothing at Doom9 about it, is it true ?? Where to download latest version ?

[Doom9]

actually.. the app will be named besweet starting from the next version and it will indeed support ssrc. But atm there's been no release so you just have to be patient (and may I suggest buying a new soundcard in the meantime ;) you know what I mean...

[DSPguru]

BeSweet v0.81 can be downloaded from the ultra mega thread (http://rilanparty.com/vbb/showthread.php?s=&threadid=7245&goto=lastpost)

[ChristianHJW]

Originally posted by Doom9
(and may I suggest buying a new soundcard in the meantime ;) you know what I mean...

:D :D Soundblaster Live! Gold 4 channel, direct predecessor of Platinum, a bit outdated but supports 48 KHz fine .... my problems are

1. my world famous company laptop on NT4, Toshiba Tecra 8100 with Yamaha DS-XG soundard .... no 48 KHz . This is the one i use the most for watching DivX movies !

2. my brothers PC, SB Live 1024 .... no 48 KHz

3. my fathers PC, 'dont ask me' soundcard .... by no means 48 KHz

4. ...... and so on

Its reall time you start believing me :

Downsampling from 48 KHz is a must !!!!

and it won't deteriorate the sound quality when SSRC is being used in 24 Bits mode.

[DSPguru]

Originally posted by ChristianHJW
and it won't deteriorate the sound quality when SSRC is being used in 24 Bits mode.

BeSweet's SSRC.dll works on floating-point. that's 32bits.

[ChristianHJW]

Originally posted by DSPguru BeSweet's SSRC.dll works on floating-point. that's 32bits

:) :) :) !!!

I just tried to download it ( really curious now ) but it won't let me :( !

[DanniDin]

Try My Page:

http://worldzone.net/computers/dannidin/

Danni.

[ChristianHJW]

... Danni, what a wonderful site !! Worked fine ... trying it now !!

[Doom9]

2. my brothers PC, SB Live 1024 .... no 48 KHz
my sblive 1024 has 48khz support. What's wrong here?

And please keep it down with the downsampling is a must.. or you run into risk of being kicked out. There's a few things in ripping where I accept no compromises and downsampling for divx is one such thing.

[pacohaas]

before we start all this bashing again, I think someone should set up a listening test. I'm thinking in a blind listening test, doom, you couldn't tell the difference between 44.1 and 48, on mp3 clips of equal size.

[pacohaas]

DSPguru said:
BeSweet's SSRC.dll works on floating-point. that's 32bits.Does this mean that you're taking 32 bit output from azid, feeding it through ssrc and straight into lame at 32 bits? i didn't think lame supported 32 bit input. It does however support 24 bit input. I believe that's why ChristianHJW and many others use 24 bit azid output and resample that in SSRC before feeding that into LAME. That being the case, perhaps your program should "just" use 24-bit calculations.

[Doom9]

I don't really care if I can hear it or not.

1) You should not let outdated hardware dictate your ripping style. All the people who stubborny refuse to go with the latest technology will one day wake up and find that they've been left behind. Towards that end being downward compatible is mostly a bad thing, not a good one. You should not carry around additional weight. I'm a big supporter for legacy-free... and downsampling is just part of this legacy. Let me ask you something... do you still buy VHS tapes just because your neighbor hasn't got a DVD player yet? Well.. I certainly don't.

2) downsampling takes time for an operation I don't need.

3) Downsampling certainly doesn't improve quality.

That being said.. discussion ended. There's no need to debate this as I'm never going to change my opinion ever and on my page there never will be a divx ripping guide that contains downsampling ever.

[ChristianHJW]

Originally posted by Doom9
And please keep it down with the downsampling is a must.. or you run into risk of being kicked out.

Where are the smilies ??? Still searching .... cant believe it ... drunken ?? Lunatic ??

There's a few things in ripping where I accept no compromises and downsampling for divx is one such thing.

O.k. big boss, listen !! As you are obviously serious about what you said above regarding 'kicking' a member out that has a different opinion than you have ..... here is my response :

If you dont apologize for what you said than there is really no need to 'kick' me ! Be sure you'll never see me again !!!

Really no need to repeat my points about why its better ever and ever agin ... you should be intelligent enough to understand that in many many cases its simply impossible to just exchange the soundcard , downsampling is taking 10 minutes longer using Danni's GUI and doesnt affect sound quality if SSRC is used !!

Waiting for your apologies .....

[DarkAvenger]

Yeah, Doom9, give it to them left and right. :D

I hate 44,100kHz. Does anybody know the story behind this odd number? It has something to do with VHS....;)

[tangent]

Originally posted by Doom9
I don't really care if I can hear it or not.

1) You should not let outdated hardware dictate your ripping style. All the people who stubborny refuse to go with the latest technology will one day wake up and find that they've been left behind. Towards that end being downward compatible is mostly a bad thing, not a good one. You should not carry around additional weight. I'm a big supporter for legacy-free... and downsampling is just part of this legacy. Let me ask you something... do you still buy VHS tapes just because your neighbor hasn't got a DVD player yet? Well.. I certainly don't.

2) downsampling takes time for an operation I don't need.

3) Downsampling certainly doesn't improve quality.

That being said.. discussion ended. There's no need to debate this as I'm never going to change my opinion ever and on my page there never will be a divx ripping guide that contains downsampling ever.

Theoretically, there should be no difference between 44.1kHz and 48kHz when you encode using LAME because most settings will be lowpassing at something below 20kHz.

Agree that downsampling takes time.

Quality is a two-sided issues. There are distortions and noise added in the process of downsampling, but at the same time, it's known that LAME has some problems with 48kHz and LAME has not been tuned for 48kHz input.

[DSPguru]

ATM, lame_enc.dll doesn't support 32bit input, but i plan to release a special version with 32bit support.
that way, the whole process (azid,ssrc,boost,lame) would acheive superior quality !

[BlackSun]

I don't really understand your point of view Doom9... Downsampling is of course bad in most case, but they are a lot of people who need to downsample their audio track. Having a old SB16 ISA for example, using a laptop where you can't change your sound card...

It's like brwosing a 1024*768 website with a 640*480 max resolution

[pacohaas]

Originally posted by DSPguru
ATM, lame_enc.dll doesn't support 32bit input, but i plan to release a special version with 32bit support.
that way, the whole process (azid,ssrc,boost,lame) would acheive superior quality ! Are you prepared to release a new version as often as say mitiok or JB? It might just be easier/and faster/and stil very high quality to do the whole thing at 24 until LAME. Just a thought, I know you're certainly capable of doing it all in 32bit so more power to you if you're up to the task of daily LAME builds.

...of course a new "stable" version is right around the corner so perhaps people won't be so "luctant" to upgrade to bleeding edge alphas for a while.

luctant=not reluctant, it's late enough to be called early so please forgive my lack of english

[MaTTeR]

I've always downsampled using SSRC because my ears can't hear the difference. It really doesn't take alot of extra time on my machines to do the 2 pass conversion. No one else listens to my RIPs so I customize them to my ear :-)

In saying that I certainly hope we haven't lost ChristianHJW from the board, especially over a subject like this. Can we kiss and make up? haha

[Rhaegar Targaryen]

Originally posted by ChristianHJW

Downsampling from 48 KHz is a must !!!!
[/B]

Hi,

I have had some engaging dialog with you in PM about audio encoding, but at this I must step in and say... That is a load of crap! ;)


Do you realize that almost all consumer audio cards are PC99-spec which means they exclusively deal with 48KHz audio? *Everything* is resampled to 48KHz internally - unless it is already 48KHz! One of the main reasons for including that in PC99 spec was to achieve best quality/compatibility when used with new millenium media such as DVD, DVD-Audio, etc. etc.

Also, the SB Live! series has a well-publicized problem with 44.1KHz that I'm surprised you don't know about. (more specifically: with the 44.1 -> 48 algorithm, I believe)

Keeping things at 48KHz is the best, for all situations, as far as I am concerned. LAME having a problem with 48KHz? Either fix the problem, or find another encoder. Afraid that 48KHz occupies too much bandwidth for a 1CD-rip's MP3 track? Just live with the 5% degradation of the video file that occurs by decreasing the bitreate by 25kbps, or teach yourself to use SBC better.

[DarkAvenger]

Ok, to be serious again, there is one point where I support Chris: Doom9 went to far with his statement and should apologize if it was meant seriously. "A line must be drawn here!" (Cpt. Piccard in First Contact ;) ) On the other hand Chris shouldn't say "downsampling is a must". Perhaps a must for him - accepted, but not for the rest of us.

I hate downsampling because of the various reasons listed here in this thread. Furthermore I don't give a damn how a rip sounds on the computer or even some crappy notebook. I only listen to it through my Denon AVR 1801 which gets its input digitally, so f### soundcards. Though DPL is a lot less nice than DD5.1, it is OK when you listen through a good amp. I want to enjoy a movie and sitting in front of a computer doesn't really cause enjoyment (not even mentioning the fan noise and small monitor in contrast to a nice 16:9 TV set...).

@Chris

Haven't you searched for better drivers. Esp. that problems with the sblive shouldn't be there. AFAIK all lives can do 48kHz. If you have enough RAM you should try w2k, as well. It has wdm drivers which make 44.1 soundcards play 48kHz material decently. (At least as good as a sblive would play 44.1kHz...)

[Doom9]

Nobody in the mood for some humor anymore? Come on.. I'm not going to start kicking people left and right unless there's no other way.

But seriously.. if you don't have to downsample you shouldn't. And if you have to.. you should consider if exchanging your soundcard isn't the better solution. I know it can't always be done (notebooks for instance) but most people have a standalone computer where exchanging the souncard takes a couple of minutes tops. And if you get a new computer these days it will most likely support 48khz anyways as today everything is destined to be able to play back DVDs which just are in 48KHz. So.. downsampling is needed less and less all the time and should eventually be become just ripping history. Even the VCD and SVCD folks usually don't downsample anymore as most DVD players don't really mind that the audio on a VCD/SVCD is 48KHz as all units have to be able to play 48KHz mp2 (required by the DVD specs).

[ChristianHJW]

Originally posted by Doom9
Nobody in the mood for some humor anymore? Come on.. I'm not going to start kicking people left and right unless there's no other way.

... i knew the smilies were hidden somewhere ;) ...

[ChristianHJW]

I guess we should stop the thread here, its leading to nowhere. I promise everybody it was the last thread here about downsampling i was participating ;) ... but let me finally answer to your post.
Originally posted by Rhaegar Targaryen
Do you realize that almost all consumer audio cards are PC99-spec which means they exclusively deal with 48KHz audio? *Everything* is resampled to 48KHz internally - unless it is already 48KHz! One of the main reasons for including that in PC99 spec was to achieve best quality/compatibility when used with new millenium media such as DVD, DVD-Audio, etc. etc.
The three reasons for Creative why to design their latest products this way were
1. profit
2. profit
3. profit
Its simply cheaper to use a DAC with fixed sampling rate ( 48 KHz ) and the DSP is onboard in any case because they need it for the fancy features they are selling their product with ....
Also, the SB Live! series has a well-publicized problem with 44.1KHz that I'm surprised you don't know about. (more specifically: with the 44.1 -> 48 algorithm, I believe)
First time i hear about that. For the DSP that is on board every SB! Live the upsampling process from 32 to 48 or 44.1 to 48 JHz should be peanuts ....
Keeping things at 48KHz is the best, for all situations, as far as I am concerned.
we both forgot to add the little word that is absolutely necessary here : IMHO !!
LAME having a problem with 48KHz? Either fix the problem, or find another encoder.
Lame doesnt have a problem with 48 KHz, not at all. In fact the 'golden ears' of R3MIX forums are convinced that 48 KHz Lame tracks are better than 44.1 KHz because of better pre-echo behaviour ... of course, these guys encode at about 200 kbps in average ...
Afraid that 48KHz occupies too much bandwidth for a 1CD-rip's MP3 track? Just live with the 5% degradation of the video file that occurs by decreasing the bitreate by 25kbps, or teach yourself to use SBC better.
There is absolutely no difference in file size between a 48 KHz and a 44.1 KHz MP3 of same quality and bandwidth ... this is obvious if it is understodd how a the compressing algorithm is working ..... the information in the individual bands doesnt change with sampling rate !

[DSPguru]

Originally posted by ChristianHJW


:) :) :) !!!

I just tried to download it ( really curious now )

well, did you test it ?
u didn't post your impression...

don't you have even one good word to say about SSRC.dll ? you know that the idea of making a dll out of SSRC is being discussed for ages..

btw,
the floating Lame code is done by now, and v0.9 works fine with it, i'm now waiting for a little help of Dmitry (http://mitiok.cjb.net/) who would compile the latest source of lame with mY code.

[DSPguru]

update :
lame_enc.dll with floating-point interface is READY, all thanks to Dmitry (http://home.pi.be/~mk442837/).

so since v0.9 we would have full floating-point process !

Stay Tuned !

[ChristianHJW]

Sh..... i downloaded the latest dll's 10 minutes ago !! A short note would be nice when you were able to upload the latest dll's ... i love the idea of having 32 bits floating point audio handling without the need to reserve 10 GB for intermediate WAV files ...

Just testing your program right now, works like a charm, i just dont have a clue why i had to use the 'BeSweets List --> MP3' instead of 'BeSweets VOB/AC3 --> MP3' but i may have missed something, so its me to blame ...

Time to advertise a new audio encoding method over at DivX.com i guess ;) ...

[DSPguru]

'BeSweets VOB/AC3 --> MP3' means :
Multiple Input -> Multiple Output.

'List --> MP3' means :
Multiple Input -> Single Output.

[ChristianHJW]

Originally posted by DSPguru
'BeSweets VOB/AC3 --> MP3' means :
Multiple Input -> Multiple Output.
'List --> MP3' means :
Multiple Input -> Single Output.

..Hmmm ... so why is only 2nd working for me ?

[DanniDin]

You Mean "BeSweet's vob/ac3 -> mp3" Does Nothing?

Danni.

[DSPguru]

the 2nd will create a mp3 with the filename you picked, the first will automaticly set mp3 filename (by the same name of input files in list, but with mp3 extension).

[ChristianHJW]

Originally posted by DanniDin
You Mean "BeSweet's vob/ac3 -> mp3" Does Nothing?

A window will pop up very briefly ( of course its too quick to be able to read something ) and than its gone. 'BeSweet List' works fine.

Also BeSweet will only from the batch window, neither 'AC3 to MP3' nor 'AC3 to MP2' with ( toolame ) will work for me.

posted by DSP Guru : the 2nd will create a mp3 with the filename you picked, the first will automaticly set mp3 filename (by the same name of input files in list, but with mp3 extension).

Maybe this is where the problem is coming from ....

Another thing : SSRC is not working !!
( See attachements )

In the command line ( see attachement from next frame ) the
--ssrc( --rate 44100 --att2 ) is clearly visible, but the process window says input and output sample rate are same, 48000 ... and the resulting MP3 is 48 KHZ indeed ...

[ChristianHJW]

Attachement nr.2 showing the command line ...

Hmmm ... is the --att 2 maybe not supported yet ??? Will try this now ..

[ChristianHJW]

Sorry, i could have checked earlier !!

It was in fact the --att 2 function that was blocking SSRC, not its working fine !!

But still only in 'Besweet List' mode. Maybe because I'm on NT4 ? DD GUI reports 'Win2k mode ' ?? But all other CLI's work fine ....

[DanniDin]

The Latest betas of BeSweet and BeSweet GUI Work for Me (Under Win2K) with --ssrc( --rate 44100 --att2 ) So Maybe You Should Should Wait for the Next Release.

I Don't Have Where to Test it Under NT4. Sorry!

Danni.

[DSPguru]

only --rate is relevant for BeSweet.

[DanniDin]

But Using --att2 Shouldn't/Doesn't Prevent BeSweet from Working Properly.

Danni.

[DSPguru]

the parser ignores all unrecognized switches.

[DanniDin]

Should I Give a Different Color (Green) to All the Supported Switches by BeSweet?

Danni.

[DSPguru]

great idea !
add a checkbox to BeSweet screen : "Color all supported switches for BeSweet".

[DanniDin]

OK :)

[ChristianHJW]

... sorry to interrupt here, but at least on my world famous NT4 laptop setting --att 2 as a matter of fact will hinder BeSweet from working . When i simply disable this switch without making any other changes it works, otherwise not ... changes can clearly be seen in the command line.
A NT4 or NTFS problem ? Simply ignore it, NT4 is about to die and even i will get Win2k now at work, so you may invest your precious coding time into something more important. Danni's GUI is running in 'Win2k mode' on my NT4 laptop. So this is just FYI ....