48-->44 Khz doubt...

[soniv]

When I encode a movie using GKnot I try to make sure I DON´T check any checkbox related to the 48-->44khz sound quality conversion. Eventhough when I open the mp3 file in winamp, the program says the file is actually 44khz. I don´t understand that at all. I only know two places where we can accidentally set this conversion on: DVD2AVI (during creation of project or in Gknot itself, when you´re seting up the last config for sound files...)and I´m sure I let them off. Can anybody help me here?? Btw, I´m using Gknot 0.26.

Thx in advance,

[MoonWalker]

What bitrate are you using for your mp3??

MoonWalker

[DJ Bobo]

You can force it to keep it at 48KHz with --resample 48
But resampling to 44,1KHz is actually always good if you're encoding to a rather low bitrate (112kbps and lower), because: less sampling rate = less compression = better audio quality

[jggimi]

My understanding is that LAME will automatically downsample to 44khz for streams with bitrates under 128kbps. I read it somewhere, a long time ago. theWEF had recommended --resample 48, to avoid sync issues, and that is what I use when I dip below 128k.

[soniv]

Thanx guys... For testing I extracted with HeadAC3he. The mp3 file is finally with 48Khz, but the godamn movie still presents sync problems.
I had one prob like this before...:( Couple days ago I was trying to encode Zoolander. I was having the same problem; tried for 200 times and gave up. Now this movie (Me, Myself & Irene) seems an impossible job... I´ve encoded 5 times with all audio bitrates: 112, 128, 160 and 192 kbp/s and all of them were out of sync after the final mux process... the _movie.avi file that Gknot creates is ok. I´ve watched almost everything and it´s not skipping like the one with audio. It´s like the audio is allways trying to get ahead of the video and the player is constantly trying to correct it (so, for brief moments it gets sync!). I´m making a 2 cds DVD rip and using vob sub 2.16 to display subtitles. It might be slowing down my computer although I´ve
never seen this kinda s**t before... Hope one of you guys can help me.

Thanx a lot for your attention,

[theReal]

If you're using Bsplayer and your mp3s are VBR or ABR, then try another player. BSplayer 0.85 seems to have a problem with VBR and ABR mp3 and is always running out of sync.
Good ol' WMP 6.4 doesn't have any problems with this...

[cjaar]

@all...

I do hv the same prob with latest .27 GK. The source audio is 48K, but the vbr mp3 shows as 44k, though the audio is synced. but i encoded SUM Of all fears, where i used 112k vbr. I see sync problem in
last 30% of the movie. Gspot says its 44k, though i didnt down sampled to 44k....
does resampling to 48 will solve the sync prob?????

Thanks
cj

[TheWEF]

lame does automatic downsampling if you choose a bitrate below 128. it's done inside lame.dll. if you check "downsampling" the downsampling is done BEFORE the data is handed over to lame. this method should be more precise and could help avoid synch problems.

so i recommend to check "downsampling" if you are using a bitrate below 128. if you are using a very low bitrate you could add --resample 44 to your lame commandline to avoid further downsampling.

wef.

[cjaar]

@TheWEF....

thanks for ur reply...
I hv a query...
when i converted the movie in dvd2avi, i get the ac3 delay as 333ms...
looks this is strange?????. I checked other posts saying 80ms is unusual, then 333ms should be just crazy????

Dvd2avi says, 29 fps, NTSC, interlaced, so i didnt check any force film option. Well when i play the vob file, there isn't any audio sync prob???
A small test i did, i hd 4 vobs, all 4 loaded in dvd2avi gave me 333msec ac3 delay, but i loded the last vob which gave only -61msec delay... any idea???

Thanks
cj

[hakko504]

333ms is much, but not impossible. And the reason you get different delays at different vobs is simply that the video/audio overlap between vobs, and since they are interleaved some audio will appear in the wrong vob compared to the video, thus giving you different delays in all vobs. This is normal and is not a problem.

[cjaar]

@gknot/dspguru...

Can i do resampling to 48 on the fly... like i added this to the existing preset: "--alt-preset 112 --resample 48"... it didnt give nay error in the log file but didnt resample it to 48????
I tried "--abr 112 --resample 48", this did work but the file size was small compared with alt preset one...

I checked the usage file of lame in doc dir.. but no example with alt-preset one...

Whats the corret cmd line for alt preset with resampling ????? or how to resample on the fly???

TA
cj

[cjaar]

@all...

--alt-preset 112 --resample 48 or --abr 112 --resample

No replies :rolleyes: :confused: :p

i think --alt-preset sounds better than --abr

Thanks
cj

[piscator]

imo, it's better to downsample to 44kHz as most soundcards do not support playback at 48kHz anyway. If you have a source at 48kHz, it will be downsampled on the fly during playback anyway. So it goes at the cost of a performance penalty without improved quality. If you use 'slow' computers for playback such as a 350MHz one, this performance penalty could be the bottleneck.

Another consideration is that MP3 has worse quality in the higher frequency domain and especially if you go below 256kbps which is usually called CD quality (which is 44kHz). So the additional quality of a 48kHZ source in mp3 is highly disputable.

Piscator

[hug0b0ss]

yes this is a very interesting topic...

Personally i always use the --resample 48 option when encoding the audio

I have downloaded some other peoples rips that have had 44khz audio
and almost every time the sound gets out of sync with the movie..

IMO 48Khz is the way to go..
but again.. everyone has their own opinion on this one...

just my two cents... Peace! :D

[theReal]

most soundcards do not support playback at 48kHz anyway. Are you sure? My Philipps Acoustic Edge can record at 48kHz, so I guess it can also play back 48kHz Sound without downsampling.

I tested the difference between lame --alt-preset standard and lame --alt preset standard --resample 44 some months ago (source was 48kHz). The 48kHz file was only a few hundred kilobites bigger than the 44kHz file (at a total filesize of about 100MB). I couldn't hear any differences and a more technical analysis with EncSpot showed that there were only minimal differences in bitrate distribution, joint stereo type distribution and block type distribution.

This convinced me not to waste any time as well as quality to resample a 48kHz source to 44kHz.

[manono]

Hi-

I think he probably should have said Some Soundcards. As time goes by, there are fewer and fewer (ISA, I think) soundcards around that don't support 48kHz. Until about 6 months ago, one of my computers had an old SoundBlaster in it, and 48kHz audio sounded real nasty. But it got replaced with an Acoustic Edge. Good card, that one.

Where you been hiding, theReal? Has theReal life been getting in the way of your visits here? We've missed you!

[theReal]

Yeah, I've been turning some things upside down in my life and I also had to move, so there wasn't too much time for the forum and for encoding. But if I'm lucky, I can soon start professional training as "Mediadesigner --Picture and Sound" (this is how I translated it from German, sounds weird in English). If I succeed getting into that professional training I can soon earn a living with everything that's got to do with electronic broadcasting, filming, editing film and video, sound recording, video format conversions, video compression, DVD authoring...

Wouldn't that be great? :)

[manono]

Hi-

I don't guess that anything you learned here will directly help you in your work. But to be able to earn your living in a field that actually interests you and to be able to do work that you actually enjoy would be great. I sure hope it works out for you. Best of Luck!

[theReal]

I don't guess that anything you learned here will directly help you in your work.I'm not so sure about that anymore :)

I've been invited by two small companies who offer professional training for the kind of job I'm looking for. Both of them mostly make promotion videos for big companies. I'm sure they absolutely know how to make great videos with nice animated graphics and stuff (which I'm not good at yet). However they were both using Adobe Premiere for most of their work, nothing more professional than most of us here use.
Then, in points of encoding, they weren't so professional at all: Both companies were making mpeg1 video mainly because that's what most customers want: videos that will play on every computer. Now, to the more horrible facts... one of them encoded the mpeg1 with LSX encoder directly from Premiere without any compression-enhancing filters. The video was for CDRom, but nevertheless they used a 352x288 VCD resolution. The next company was encoding one of their mpeg1 projects with FlaskMPEG on a PIII 500.

I wasn't in the right position to critizice anything, but haven't they ever heard of CCE or TMPEG, or avisynth, or what???

I hope I get the job training at one of the companies so I can start enhancing their mpeg1 compression techniques...

[wingphil]

just to add something i've noticed (not that i'm an expert or anyhing) but a lot of my encoded movies has sync problems when played in winamp3, but play fine in other players.

also one (metropolis) froze a couple of times in all player i tried, (including divx player), except the core media player, which seems sharper and smoother somehow as well. i reckon the code is tighter, and apparently it has divx5 optimisations.