(ST=sound track..)
So look, what i did is played the AC3 file i ripped from the VOBs (448kbps 5.1 channels) in winamp using Valex's AC3 decoder for Winamp (based on AC3filter), then decided to do the crazy thing which is convert it to wav, stereo, 44.1khz. (used Disk Writer plugin, normalized the wav then), so that i'll be able to convert it to MPC then. Sure, there may be a simpler way but i didn't care..
In short, i encoded it to MPC then using the latest alpha, 1.15f, --quality 5 --xlevel. And i have a 90:00 minutes MPC file which is 84.1 MB (88,243,052 bytes) in size, 130.7kbps avarage. MPC shows it's great VBR abilities in this soundtrack very well. At points it's as low as 80kbps (and still sounds just like the original) and at points as high as 200+kbps.
In short, MPC is the ELITE! Just wanted to share this with you .
No offence at all, but Ogg Vorbis would never be able to satisfy me with such amazing sound quality at an avarage bitrate of 130.7kbps.
MPC is just great, this is another example that blows away myths that MPC is good only for high bitrates, or medium!

edit: oh because of all the excitement i forgot to mention how much i'd love an MPC directshow filter and a container that supports MPC to combine it with the video.

Can you try again, but with --quality 5 --ms 15 ? There was some times ago a thread about surround ability of mpc (and better results with this switch). Just to have an idea of the bitrate increase. Thanks

PS : maybe 1.14 will give you lower bitrate.

Just did some testing with a 407Mb audio file

--quality 5 --ms 15 = 26.7Mb
--quality 5 = 24.4Mb
--quality 5 --xlevel = 24.4Mb

Does that look correct?

I forgot a very important thing ... i made the horrible mistake which is converting it to 44.1khz wav straight with winamp, which sucks of course, since there's no filtering... So converted it to 48khz wav, then converted to 44.1 with filtering, which took loooong of course. Encoded with the same settings, and it's 133kbps.
Now, i also encoded using --ms 15 (man i think that's way too high) and it's 157kbps. Anyway, i really don't notice a difference, maybe if i'll try i'll notice, but the default is good enough for me in this case.
About 1.14, yeah i figured it'll give me a lower bitrate too, but no way, 1.15f just has too important quality improvements to give it up for a lower bitrate .
You're welcome .

Yep looks correct .
It all depends on the contents. Probably the sounds are not complex at all so MPC's able to encode at low bitrates and maintain good quality. Or there is much silence, or all together.
What's the files' avarage bitrate? You can check using file info in foobar2000 as well as Winamp.

Hi,

I was just encoding my TV rips soundtrack. using Vorbis -q4 to encode the track.
Now tested with mppenc 1.15f -q5... and here are the results

45Mins 40secs, PCM, 44100Hz, 2 Channels, 16bits

original: 460 MB (483 336 033 bytes)
vorbis: 32,5 MB (34 179 024 bytes)
mpc: 34,1 MB (35 800 440 Bytes)


and again mpc kicks ass
Avarage bitrate was 104.5.kbps.

But no way to use with Xvid

To make sure, i encoded it with version 1.14, using "--quality 5 --xlevel" The avarage bitrate is 131.3kbps. So seems that there really isn't much kbps difference from 1.15f in -this- case. (1.14 is 131.3, 1.15f is 133). About quality, don't know, didn't check, anyway my guess is that it'd be too hard to find a difference.

Thanks.
But why convert it to 44100 hertz ? MPC support 48 Kz input & output file. Is bitrate really higher ?

For --ms 15, it seems to be be difficult to hear or abx the benefits of the switch. But maybe with a Dolby ProLogic II or similar technologie (emulation of multichannel with a stereo file), the results are audible. I don't know and can't test it.

Thanks again.

Well since i don't believe in 48khz...i think it's a total waste.
A human can not hear frequencies over 22,050hz (i can't hear over ~20,000), and well, 44,100 provides 2 22,050 channels, all that's needed, so i see no need in extra -3,900- (!) ...
Here i'm just talking about playback, not editing, not pro stuff, not bits (24, 32), so for listening, 44,100 is just right, so that's why it's so important to me to use 44.1khz .
And you're welcome again .

edit: about bitrate being higher, i really don't know, didn't check and don't think i ever will..

Have you tried to encode it using --quality 8? If --quality 5 is giving such a low bitrate, who knows if even BrainDead is just about 200-224 crappy kbps? just wondering

Looks likes an overkill to me.

I won't ever do that because actually, the AC3 is not very high quality anyway, most of the time a stereo AC3 is either 192kbps or 224kbps. So, i don't like the idea of encoding an MPC with such a high bitrate. Hard to explain i guess, but maybe you know what i mean.. I consider an avarage bitrate of ~130, 140, and maybe 150 very reasonable for a stereo MPC transcoded from AC3 (be it stereo, or 5.1 converted to stereo).

OK waiting for --braindead --xlevel to finish...
Finished, avarage bitrate is 207.9kbps. Just for you ..
Gonna delete it now .

Now tested with --radio:

26,4 MB (27 757 220 bytes)

bitrate was 81.0 kbps

--radio aint tuned so much. Can't say nothing about the quality...maybe some else could..

Yeah, I understand why you want a low bitrate about 130kbps, as the fact is reduce it, not have the same bitrate....
I know braindead is severe overkill, but if even at this setting it gives such a low bitrate, the fact then is that AC3 content is extremely easy to encode for MPC.

Try also mp3 --alt-preset standard. If it doesn't give a low bitrate, then probably the fact is that MPC's subbander just beats out anything. >_<

What are you talking about?

Musepack's psychoacoustics will know when to encode high frequencies and when to omit them. Just because the file is sampled at 48KHz doesn't mean you're all frequencies up to 24,000Hz are going to be encoded. Hell, you could take a 192KHz-samplerate wavefile and lowpass that at 6KHz, or whatever. My point is, your argument about not needing high frequencies has nothing to do with the samplerate in this case.

It's true that a file encoded from a 48KHz source will likely be a bit larger than one from a 44.1KHz source, but if you avoid converting, you will have a better quality result. The 48 -> 44.1 samplerate conversion is going to cause some quality loss, and it will probably have to do another 44.1 -> 48 when it goes through the soundcard during playback.

So why not just keep it at 48KHz?

Oh, and while I'm on the rampage... I'm not sure how the Valex AC3 plugin handles 5.1 -> 2.0 conversion, or how it does the normalizing, or what kind of compression/limiting it uses, but I would make sure that it's properly mixing LFE and center channels and make sure the other settings make sense. I would still use Azid (or BeSweet, or something of that nature) to do a 2-pass normalization. As I understand it, AC3 stores the amplitude as floating-point, so you have to go through the whole file and find the peak, and then scale the whole file when decoding to fixed-point WAV. Otherwise you're re-quantizing or something when you normalize(generally considered a bad thing). The effects of all these settings might be near-negligible, but then again we're encoding to Musepack --standard, not MP3Pro; we want the best possible quality, with fewest possible compromises!

I understand that you were just doing a test to see what kind of filesize Musepack would produce, but if you want to get nitty-gritty about its quality and its size, you should perform the test under optimal conditions and see if the bitrate differs from your original report.

But even after all my yelling, I still think it's great to see Musepack performing so well on a movie soundtrack. If the filters were available, I'd certainly use Musepack for the audio compression on my DVD-rips.

(same file etc, etc)

now with --aps:

41,3 MB (45 201 868 bytes)

bitrate was 131.

So larger file size:P

and VERY slow to encode...

(Still waiting for that filter....)

I'm looking to try mpc for a movie soundtrack but didn't find the tools anywere ! for example , how (what tool ?) do you mux the mpc with the divx ? were can i find the direct show filter for wm ?

thanks!

You could maybe mux to ogg container but you have no way to decode it becouse theres no ddshow filters etc...

(Still waiting )

I was wondering if MPC will ever support more discrete channels? Maybe 255 ala ogg vorbis? And i also would like mono mode, just for correctness. And maybe full discrete mode in multichannel without any correlation savings also for multi language purposes, unless the metaformat can handle multiple audio streams. Oh i guess the direct show thing could be needed as well. Im thinking in something like ogg/ogm/mcf or whatever non avi/mov open source metaformat gets more popular.

Apparently SV8 is supposed to add this.

You have a problem of some sort with me?...
No shit.. of COURSE frequencies won't be up to 24,000hz (!)
My argument about not needed high frequencies does have something to do with the samplerate in this case. Since 48,000hz is NOT, again, NOT needed, i convert it to 44,100hz.
"but if you avoid converting, you will have a better quality result."
You are wrong. I lose absolutely no quality when i do a very high quality conversion from 48khz to 44.1khz. If you use cooledit, then convert using pre/post filter and quality 999 (which is of course not needed, ~600 is more than enough) and try to find a difference, you will never find a difference.
"and it will probably have to do another 44.1 -> 48 when it goes through the soundcard during playback."
That is, if you have a resampling soundcard...
"So why not just keep it at 48KHz?"
Because it's not necessary. And i don't like the idea of using unnecessary things. That's my opinion, since it's not yours that's fine, use 48khz.

The Valex AC3 plugin handles 5.1 ->2 conversion very well as far as i could hear, it's very accurate. And of course i DID make sure it mixes the LFE and center channels properly, and i always make the "other settings make sense"..
I was not just doing a test without making sure the quality is the best it can be. I'm just not that kind of person.

Used Dibrom's latest 3.90.2 ICL compile, encoded with APS. Took a looooong time... (i just love MPC's speed)
The avarage bitrate is 155.4.
I'm not gonna try with the latest 3.94 alpha, it takes long , and it's not so tuned right now anyway.

5.1 AC3 ---> Stereo wave / Ogg / mp3 / mp2 ====> fastest and best tool is: HeadAC3he

believe me, try it.
for samplerate conversion there is ssrc.dll afaik (but leaving 48 kHz as it is, I prefer....)


http://darkav.de.vu/

I know the tool, i've been using it some months ago. Didn't notice it supports MPC..

edit: ah no, indeed it doesn't support MPC..

Okay, anyone knows if there's a 'dumb' - no muxing - mpc directshow filter?

As far as i know no MPC DS filter exists...

http://www.hydrogenaudio.org/forums/index....=ST&f=12&t=4440

The search function is always your friend ... 1st posting in MPC-tech section.



Sorry mate, Franks wish is our command here, and we all have to admit it makes sense also to wait for SV8 first ... development freezed on MPC support in matroska for now ...

Sure that's perfectly fine . Waiting.