BSN=BeSweet+Nero

[JasonFly]

Thanks for your reply.


@calinb:
The sample rate tab in CoreAAC show 22050-->22050 in The core media player but show 22050-->44100 when mkv is played with Media Player Classic.That's really strange but that works.
I'll try the latest CoreAAC(I thought I had the latest) and I'll see.


@Tuning:
That could have been an explanation but my file is approximately 43 min long.
Does it do the same thing in cbr mode?

[calinb]

Originally posted by JasonFly
@calinb:
The sample rate tab in CoreAAC show 22050-->22050 in The core media player but show 22050-->44100 when mkv is played with Media Player Classic.That's really strange but that works.
That is really strange. I've never used the core media player--any chance that it's not using the same filter that MPC is using? It seems like it would be using the CoreAAC filter, but maybe it has an internal filter.

The reason I ask is usually, when you see 1/2 the HE sample rate, it means the player is playing back the AAC as LC instead of HE. I guess you could consider it a backward compatiblity mode for LC-only players. It appears that core media player thinks it's LC. Have you listened to some music with a lot of highs, like cymbals, on some good speakers or headphones? If it's playing back as LC, it should kill your highs, I think. Maybe a listening comparison between core media player and MPC is in order.

[JasonFly]

Yes, that should be something like that.MPC show AAC+SBR in the profile tab.

I'll try to find info about The core media player and i'll try a little test with music samples like you suggest.
Thanks.

EDIT:
I have tried with the latest version of the CoreAAC filter(5th october, mine was 7th august) and The coremedia player show now 44100-->44100 in the sample rate area and AAC-LC in the profile area.
MPC show 22050-->44100 and Reserved for the profile, that's better.

[Aboo]

Originally posted by calinb
I was just wondering whether 44.1 or 48 is better overall, if you have a 10kx chip. (Nero is best at 44.1 vs. 10kx chip is best at 48--at least according to theory and urban legend :)) tiki4 is probably correct that we won't hear the difference. I haven't tried any listening tests for this.
To tell you the truth, i never noticed any difference in sound quality between 48 and 44.1 I cant say that i have a "that bad" sound system. I have 4.1 Cambridge SoundWorks FPS1600 - the maximum for my SBLive! Value. Maybe i'm just dumb-eared :D ;)

[Aboo]

Originally posted by JasonFly
I have tried with the latest version of the CoreAAC filter(5th october, mine was 7th august) and The coremedia player show now 44100-->44100 in the sample rate area and AAC-LC in the profile area.
MPC show 22050-->44100 and Reserved for the profile, that's better.

Just a suggestion, but did those of you who had problems turned on "AAC is HE-AAC, SBR, AAC+" flag in mkvmerge?

[calinb]

Originally posted by Aboo
Just a suggestion, but did those of you who had problems turned on "AAC is HE-AAC, SBR, AAC+" flag in mkvmerge?
mkvmerge 7.0 introduced HE identification of mp4 files so you don't need the flag with HE-AAC in mp4. However, you still need it if you mux an aac file.

[tiki4]

One hint about Nero AAC encoder:

Nero chooses automatically LC if you try to encode a stereo file with a higher preset than streaming. So for a stereo file you just have to select streaming preset and HE profile and you get what you want.

The same goes in principle for 5.1 input files. You can select HE up to streaming preset, anything else gets encoded as LC.

For CBR encoding you can select HE up to 96 kBit/s and up to 128 kBit/s for a 5.1 source.

tiki4

[Stranger]

I've a strange problem....
I'm using Besweet + bsn.dll + AAC.dll (and others) from nero 6.0.0.11 so I use the -o in azid for correct channel mapping and -6chold in bsn.

The command line is the following:

BeSweet -core( -input test2.ac3 -output test2.mp4 -logfilea BeSweet.log ) -azid( --maximize -c normal ) -azid (-oc, l, r, sl, sr, lfe) -bsn( -6chold )

Here is the log from besweet:

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : test2.ac3
[00:00:00:000] | Output: test2.mp4
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 13.822dB, Compression: Normal
[00:00:00:000] | LFE levels: To LR -INF, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[00:08:13:216] Conversion Completed !
[00:01:54:000] <-- Transcoding Duration


From the aac panel I simply choose a preset (despite it is he or lc, cbr or vbr... I tried almost everything)
As you can see the input test is a 6-ch ac3 but the output file is a 2-ch file and really I don't know why :(
If I try to use the file the CoreAAC DS tells me it is a 48.000 khz but only with 2->2 channel
But why?
Where is the error?
Tnx a lot.
Stranger.

[calinb]

Originally posted by Stranger
Where is the error? I tried your command line and it worked. What versions of the BeSweet/BSN/azid tools are you using? You really need the latest. We need to see the top of the logfile too. Here's mine:

BeSweet v1.5b22 by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using bsn.dll v0.2 by DPeshev,Richard,DSPguru (http://DSPguru.Doom9.org).

Logging start : 10/13/03 , 13:39:04.

C:\Program Files\Gordian Knot\BeSweet.exe -core( -input G:\RTV\MP4 Tests\vobtest AC3.ac3 -output G:\RTV\MP4 Tests\vobtest AC3.mp4 -logfilea G:\RTV\MP4 Tests\BeSweet.log ) -azid( --maximize -c normal ) -bsn( -6chold )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : G:\RTV\MP4 Tests\vobtest AC3.ac3
[00:00:00:000] | Output: G:\RTV\MP4 Tests\vobtest AC3.mp4
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:000] | Total Gain: 3.509dB, Compression: Normal
[00:00:00:000] | LFE levels: To LR -INF, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[00:00:04.426] W7: Downmix overflow (2: +0dB)
[00:01:00:000] Conversion Completed !
[00:00:31:000] <-- Transcoding Duration

Logging ends : 10/13/03 , 13:39:35.

BTW, you don't need the -azid (-oc, l, r, sl, sr, lfe) option. -bsn( -6chold ) is sufficient.

[DSPguru]

i released a gui for transcoding to vorbis & aac, it's called The OagMachine and it can be found on my stable branch @ my website.

enjoy!

[Tuning]

i released a gui for transcoding to vorbis & aac, it's called The OagMachine and it can be found on my stable branch @ my website. enjoy!

Thanks DSPGuru!Now AAC encoding became even easier.

[calinb]

Originally posted by DSPguru
i released a gui for transcoding to vorbis & aac, it's called The OagMachine Yeah, I downloaded it this morning and it's great! :D Thanks again DSPguru!

[Hobojobo]

what is the problem.
I try to transcode a ac3(5.1) to aac(5.1) with The OagMachine.
I end up with error configuring bsn!

I tried all all listed channel modes.
Nero 6.0.0.19 (Demo) is installed.


Here is the log:


BeSweet v1.5b23 by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using bsn.dll v0.2 by DPeshev,Richard,DSPguru (http://DSPguru.Doom9.org).

Logging start : 10/23/03 , 23:27:47.

C:\OagMachine0.1\BeSweet.exe -core( -input C:\audio.ac3 -output C:\audio.mp4 -logfilea C:\OagMachine0.1\BeSweet.log ) -azid( -g max -L -3db -c normal ) -bsn( -6chold -config -path C:\Programme\Gemeinsame Dateien\Ahead\AudioPlugins ) -profile( The OagMachine v0.1 )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : C:\audio.ac3
[00:00:00:000] | Output: C:\audio.mp4
[00:00:00:000] | Floating-Point Process: No
Error 84: error configuring bsn!
Quiting...
[00:00:00:000] Conversion Completed !

Logging ends : 10/23/03 , 23:27:53.


Can anybody help me out?
Thank you.

[Hobojobo]

Stupid me.:o
First read then think then speak (post).
I sometimes get mixed up, what is first...

Copying all the needed files into the oagMachine folder, solved the error, as mentioned before.
I checked 'Set plug in path' and even entered the right path , though. :confused: That should have had the same effect, shouldn't it ?

[tiki4]

Hi,

just wanted to say congrats to DSPGuru for his new tool. Excellent work.

Now that everyone can use that tool, can someone share his experience regarding quality of 5.1 AAC?

I encoded a song from Springsteen's NYC-DVD some days ago to HE-AAC 'streaming' preset and the sound is still superb. But now I did Star Wars Episode Two and somehow the sound generated from the 448 kbps AC3 sounded rather terrible. I don't know why as I used the same settings with Nero 6.0.0.19 dlls and downsampled in BeSweet to 44.1 kHz.

Any ideas someone?

tiki4

[Tuning]

Do you checked the Core AAC filter is showing the same "Reserved"state in both tries.Sometimes i found even if you select HE-AAC the encoder uses LC AAC.

-Tuning

[tiki4]

Yep.

Both times CoreAAC shows Reserved and 22050->44100.

It's not so much a loss in high frequencies but the 'surround feeling' of the AC3 of Star Wars is somewhat reduced and also somehow the orchestral music is less pronounced. Dunno how to describe that properly. I think the music on the Springsteen DVD is more or less stereo coming from the front speakers, while the Star Wars DVD is very pronounced surround sound. Maybe that stresses the encoder too much. Of course Star Wars produces smaller bitrate (~160 vs. 180). This is something I already know from LAME (try lame -aps with music from CD and with a stereo movie soundtrack, the first ends up around 190 - 200 kbps, normal movie with many dialogues ends up around 160 - 170).

tiki4

[bond]

Originally posted by tiki4
I don't know why as I used the same settings with Nero 6.0.0.19 dlls and downsampled in BeSweet to 44.1 kHz.hm, try it without the downsampling in besweet as nero also downsamples automatically when using sbr (so you are downsampling twice i think)

[tiki4]

Nero 6.0.0.19 has issues with sampling rates other than 44.1 kHz. See above in this thread. (Otherwise, I tested with 6.0.0.11 aac.dll and it sounded the same). I will try without HE- now.

tiki4

[StoneRoses]

Just want to confirm that if I used Nero AAC Codec version 2.5.5.8 (come with Nero 6.0.0.19) for encoding 5.1 AAC and playback with latest CoreAACDs or foobar2000 latest mp4 plugin found at:
http://www.hydrogenaudio.org/index.php?showtopic=6428&st=100&hl=

I should use switch -bsn ( -6chnew ) instead of -bsn ( -6chold ) to get correct channel mapping.

Everybody here seems to use -6chold, so I'm a little bit confused.

[tiki4]

If you use the dll's from Nero 6.0.0.19 (aac.dll, aacenc.dll and NeroIPP.dll) use -6chnew. Only if you use 6.0.0.11 or you change aac.dll to the 6.0.0.11 version you have to use -6chold.

aac.dll seems to be the one that handles the input. The reason some people switch aac.dll to the older one is that the new aac.dll from 6.0.0.19 has issues with 48 kHz input.

tiki4

[StoneRoses]

Thank you. :)

[menno]

Originally posted by tiki4
the new aac.dll from 6.0.0.19 has issues with 48 kHz input.

It does not. The application you use has trouble correctly feeding data to the plugin (no correct resampling).

Menno

[bobsc]

Originally posted by menno
The application you use has trouble correctly feeding data to the plugin (no correct resampling).
Could you elaborate?

[menno]

Sure, sometimes the AAC configuration can determine that the audio data needs to be resampled. The application using the plugins should handle this by resampling the data.

Menno

[bobsc]

Originally posted by menno
Sure, sometimes the AAC configuration can determine that the audio data needs to be resampled. The application using the plugins should handle this by resampling the data.
Sometimes? I thought you should always resample to 44.1 kHz with current Nero.

[menno]

Depends on the rest of the configuration.

Menno

[bobsc]

@Menno
Could you please clarify?

[menno]

If the bitrate gets lowered the samplerate will also get lowered, to give the best quality result.

Menno

[bobsc]

@Menno
So, at what bitrate should you resample to 44.1?
BTW, welcome to the forum.

[Tuning]

Well,I did a small test on account of your consideration menno,encoded a small mp3 file(48Khz) in nero wav editor(with 60019 AAC.dll) and tried to play in MPC.But still MPC crash.(Same as happened in BSN,thus I think there is no problem to BSN regarding feeding encoder). Thus it feels that only problem is the 3ivX splitter.

[menno]

Originally posted by Tuning
Well,I did a small test on account of your consideration menno,encoded a small mp3 file(48Khz) in nero wav editor(with 60019 AAC.dll) and tried to play in MPC.But still MPC crash.(Same as happened in BSN,thus I think there is no problem to BSN regarding feeding encoder). Thus it feels that only problem is the 3ivX splitter.

Well, yes, that is ALSO a problem :)

But this is a problem in the 3ivx splitter. They fixed it in recent builds, dunno if that is released yet or just beta.

Menno

[Tuning]

I need to know how the HE/LC switch is made depending the bitrates.I.e when is exactly HE used and when LC.(Some times even if you use HE ,we get LC as the encoding).Both in 2.0/5.1 situation.10X.

[menno]

Hmm, that's hard to say :) I don't make the presets. But when using CBR I think HE is used from 80kbps and lower.

Menno

[StoneRoses]

[off-topic]
This is a bit off topic. :)
My recommend software for playback AAC (and every other audio formats) is foobar2000. FB2k is the excellent music player for PC. With right component, it can play almost every auio format including multichannel AAC in both mp4 container or just plain AAC stream (.aac). IMHO, FB2k is the MPC for audio world.

Links
Official Website http://www.foobar2000.org/
Case's Special Installer (recommended) http://www.saunalahti.fi/~cse/html/foobar.html
Foobar2k Forum http://www.hydrogenaudio.org/index.php?showforum=28

UI Customisation
Main windows - foobar2000 formating http://pelit.koillismaa.fi/fb2k/index.php
GUI - foobarlooks http://www.barciaonline.com/aural/foobarlooks/foobarlooks.htm

[/off-topic]

P.S. Thanks DSPGuru for making a very good transcoding utility.

[Tuning]

@StoneRoses

Thanks for the links.
BTW,I have a doubt.Do Foobar plugin play mulichannel files with correct channel mapping?

[StoneRoses]

Yep. AFAIK, Both Foobar mp4 component and CoreAAC use FAAD2 for decoding AAC. The channel mapping on both software should be same.

You can get latest foobar mp4 plugin and latest CoreAAC at this thread (john33's binary)
http://www.hydrogenaudio.org/index.php?showtopic=12915&hl=foo_mp4

Note: I suggest you to use foo_mp4.dll that come with case's special installer. It use FAAD2 code plus new mp4 tagging routine that fixed tagging bug. (I used AAC for compressing music to use with Apple's iPod (sync via iTunes), so mp4 tagging is very important for me)

[Tuning]

Thanks again.
I think the Winamp plugin is still incorrect,regarding channel mapping.

[Tuning]

Some more simple 5.1 HE/LC tests again.The new foobar and CoreAAC plugins are correct in channel order if we use the 6chold switch in BeSweet.
Winamp can correctly play in 5.1 if the Switch is 6chnew.
So again as expected the Winamps channel mapping not in order of requirement.

#Edit#
Oops....
Note:All tests besed on 60011 AAC.dll.Right Gaia?

[Gaia]

Originally posted by Tuning
Some more simple 5.1 HE/LC tests again.The new foobar and CoreAAC plugins are correct in channel order if we use the 6chold switch in BeSweet.
Winamp can correctly play in 5.1 if the Switch is 6chnew.
So again as expected the Winamps channel mapping not in order of requirement.

@ Tuning

If you use latest .dll's use always 6chnew. If you use old 6chold is right. Read Menno's answear about this "48 khz problem".

What versio of Winamp are you using? If you're using Winamp3 it's mp4 plugin is old and outdated. It's not going to be updated anymore.

Also cheack out this thread in Hydrogenaudio http://www.hydrogenaudio.org/index.php?showtopic=14647&

Edit: You're also confusing some people because you don't say what .dll's you're using...

[Tuning]

I could understand the Menno's replay on 48Khz problem,and it is only to be with 60019 AAC.dll.
And sorry for generating confusion.what i was trying to do was to find correct channel mapping in WinAmp no matter what version of AAC plugin you use.Finally what I found is that you can only get correct channel mapping in winamp iff you use Old(60011)AAC.dll and set switch 6chnew.

When using new AAC.dll(60019)in no way I could find correct channel mapping both in 6chnew and 6chold switch.So what i was trying to bring-up is that the in_mp4 plugin for Winamp 2.9x/Winamp 5.0 is still incorrect regarding channel mapping.So if you are using new dll(60019) a new channel setting will be required or otherwise correct winamp's plugin.

For foobar and CoreAACDS,you quoted is right.To simply overcome the 48khz problem we need to downsample to 44khz while using new AAC.dll(60019).But still the down sampled file will not be playable in DS Media players untill the corrected 3ivX splitter is available.

BTW,I have already downloaded the latest plugins from the link provided by StoneRoses.

Hope everything is right....:rolleyes:

Bye:)

[sillKotscha]

sorry guys, this time I'm really too dumb to get it to work...

what exactly is needed??

- I have Oagmachine
- latest BeSweet
- and the bsn package (btw, the dlls in the package are different from latest BeSweetpackage - which dlls are preferred????)
- and of course Nero (v. 6.0.0.15)

I define the path in Oagmachine to the Ahead/Audioplugins but bsn config won't open...

ok, what shall I do???

1. have the Oagmachine folder
2. put in latest BeSweet with all corresponding dlls
3. put in bsn.exe
4. and last but not least define the Ahead_plugin_path

is this the way to go???

thanks in advance - Sill

[bobsc]

Originally posted by sillKotscha
ok, what shall I do???

1. have the Oagmachine folder
2. put in latest BeSweet with all corresponding dlls
3. put in bsn.exe
4. and last but not least define the Ahead_plugin_path
3. You don't need. 4. Doesn't seem to work, so you have to put the Nero dlls in your folder.

[sillKotscha]

ahh, thanks :rolleyes:

and one last: which nero dlls exactly??? only 2 are used, am I right???

there is a special wav_replace floating around for correct channel order in Nero concerning multichannel encoding... does it have to be in Oag_folder as well, or just stick with the nero one??

thanks again :)

[bobsc]

Aac.dll, aacenc32.dll, neroipp.dll that's it.

[sillKotscha]

thank you Sir !! :)

[filewalker]

But still the down sampled file will not be playable in DS Media players untill the corrected 3ivX splitter is available.
@Tuning
What you can try is to load the downsampled AAC file(.mp4) in MKVMerge and save it as MKA. So the MatroskaSource-->Matroska Splitter --> CoreAAC --> Audio Renderer come in use instead the 3ivX Splitter.
This works for me if I play back a 2.0 stereo HE AAC file(encoded with Nero 6.0.0.19).

Now I encoded a 5.1 with HE AAC with Oagmachine and saved it in Mkvmerge like described above as MKA. Playback works fine but I only have a 2 channel amplifier..so I can't test if the channel order is correct...

Maybe it works in this way...

edit: only CoreAAC Decoder works with MKA but not the NeroDigitalAudioDecoder

Cu filewalker

[Tuning]

Originally posted by filewalker
What you can try is to load the downsampled AAC file(.mp4) in MKVMerge and save it as MKA. So the MatroskaSource-->Matroska Splitter --> CoreAAC --> Audio Renderer come in use instead the 3ivX Splitter.

Thanks filewalker,I have never thought of this.I will try.All my tests were done with bare mp4 files.So this sounds to be working.Thanks again.
#EDIT#Playback works fine but I only have a 2 channel amplifier..so I can't test if the channel order is correct...
Channel order is as posted by Gaia.Use 6chnew if you are using new AAC.dll(60019),other wise use 6chold(60011).

-Tuning

[StoneRoses]

Tunning,

Mp4 file encoded with newer Nero has new feature for gapless playback (very important for live Music CD/DVD rip). Unfortunately current version 3ivx splitter (and Apple's iTunes) don't like it.

Here is another easy way to overcome 3ivx mp4 spliter bug. But you will lose gapless information (not so important for movie and music wip gap between track).

1. Open the file in FB2k.
2. Right click the track in main windows and choose "Extract AAC track" you will get .aac file.
3. Open the AAC file in foobar and then right click and choose "Convert to MP4"

The resulting mp4 file (no arbitary frame size) now can play fine in MPC using 3ivx splitter.

I don't know exactly about the incorrect channel mapping in Winamp, becuase I don't use it (FB2k is a lot better in my opinion). But if you encode the file using tool/parameters that produce correct channel mapping when playback in Winamp, your file is not a standard 5.1 AAC file anymore.

[Tuning]

Thanks for the comments Stone Roses,any way I'm going to try filewalker's idea.If it still not working then I might wait for the next version 3ivx splitter.
Bye.