Converting 5.1 FLAC and DTS

[sarofski]

Hi guys,

I want to convert 5.1 ch FLAC/DTS audio files to 5.1 AAC/AC3.
Is there any free software for that?

Thanks

[El Heggunte]

LameXP or Hybrid.

If you're not afraid of the command-line interface , eac3to + qaac / fhgaacenc.

[ozok]

You can use TAudioConverter too.

[sarofski]

Thanks

I tried TAudioConverter and LameXP.

TAudioConverter: I really liked this program. It works perfect for AC3 but for some reason it failed for all the 4 AAC encoders it offer ( It's so weird, cause the 1st time I installed it, 2 of them worked, but when I updated to the new version, none of them works anymore and when I turned back to the old version again none of them works!! ).

LameXP: Works fine for the only AAC codec encoder it has ( NeroAAC ) but it doesn't cover AC3 format.

So I guess I can get what I want for FLAC files by using these 2 programs!

But none of them could encode DTS 5.1 Ch to AAC/AC3 5.1 ch. so any program for it?

Thanks again guys

[ozok]

Are you sure you have quicktime for qaac and winamp for fhgaacenc installed? Try sending logs to see what happened.

[hello_hello]

I use foobar2000 for audio conversion quite a bit. Mainly because you can load a bunch of audio files into a playlist and it'll convert as many simultaneously as you have CPU cores until it's done, or you can convert to multiple formats at the same time. You'd need to download the relevant encoders manually, and possibly the DTS decoder plugin from the foobar200 site.

Alternatively, you could just use MeGUI's audio encoding section.

[sarofski]

Are you sure you have quicktime for qaac and winamp for fhgaacenc installed? Try sending logs to see what happened.

Well I do have winamp and quicktime installed, although I think these(Encoders) come with the installation, Since the only missing encoder seems to be "NeroAac" which I downloaded and put in the folder. But even the Nero encoder doesn't work!!

[sarofski]

BTW just in case I manage to save the problem for TAudioConverter, Which AAC encoder is better?

FAAC
QAAC
fhgaacenec
NeroAacEnc

[sarofski]

I use foobar2000 for audio conversion quite a bit. Mainly because you can load a bunch of audio files into a playlist and it'll convert as many simultaneously as you have CPU cores until it's done, or you can convert to multiple formats at the same time. You'd need to download the relevant encoders manually, and possibly the DTS decoder plugin from the foobar200 site.

Alternatively, you could just use MeGUI's audio encoding section.

Thanks, I'm gonna try them.

[hello_hello]

At a high enough bitrate I don't think there's much between the AAC encoders, if anything. Every single comparison test I've read only compares them at bitrates lower than you'd be likely to ever actually use, I assume because any comparisons at higher bitrates would be very unlikely to produce reliable results.

I use the Nero encoder myself as both foobar2000 and MeGUI have presets for it. I just use the default quality setting of Q0.50 which is probably comparable to LAME's standard V2 VBR preset.... well at least in respect to the file sizes it produces. I use the same quality setting whether I'm encoding stereo or multichannel audio.

I'm pretty sure foobar2000 doesn't have an AC3 encoder preset so you need to make your own. Once in the encoder setup section you'd add a new encoder and choose, "custom", then point foobar2000 to wherever you've put the AFTEN encoder. For the command line, this works fine with foobar2000. The -b parameter being the bitrate, which you can of course change. If you don't specify one, Aften picks it's own. http://aften.sourceforge.net/longhelp.html

-readtoeof 1 -v 0 -b 448 - %d

And one you get to know foobar2000.... which admittedly can take a little time..... you'll probably realise it's a very good audio player.

[ozok]

In settings window you must specify NeroAACEnc's path. I use QAAC generally, but apart from FAAC, rest is similar and really good. Also what happens when you set "Bit Depth" to 16?

[hello_hello]

Which settings window? Which program? Generally if you select a bitdepth for a lossy encoder I think you'd be specifying the maximum input bitdepth. Lossy compression doesn't have a fixed bitdepth, so logically I'm not sure what else it could be aside from the input bitdepth. If you selected 16 bit while the source was 24 bit etc, I assume it'd be downsampled to 16 bit first, then sent to the encoder. That's my best guess, at least.

[ozok]

I was talking to OP about TAudioConverter. Also when you specify a bit depth in TAC, FFMpeg decodes source to that bit depth. Also QAAC and fhgaacenc works here even if source is 24bit so I'm guessing something else is broken at OP's end.

[pascal]

For conversion of DTS 5.1 to AC3 5.1 eac3to works fine

[sarofski]

In settings window you must specify NeroAACEnc's path. I use QAAC generally, but apart from FAAC, rest is similar and really good. Also what happens when you set "Bit Depth" to 16?

I have specified the NeroAACEnc's path and I set Bit Depth to "Original".

I just found out that FAAC works when I set AAC Extension as "MP4" or "M4A" but not with "AAC" and "M4B"

But for "QAAC" , "fhgaacenc" and "NeroAacEnc" it wont work no matter what I do! ( Changing Bit Depth/AAC Extension, ... )

BTW is there anyway to convert DTS files by TAudioConverter?

Thanks

[sarofski]

At a high enough bitrate I don't think there's much between the AAC encoders, if anything. Every single comparison test I've read only compares them at bitrates lower than you'd be likely to ever actually use, I assume because any comparisons at higher bitrates would be very unlikely to produce reliable results.

I use the Nero encoder myself as both foobar2000 and MeGUI have presets for it. I just use the default quality setting of Q0.50 which is probably comparable to LAME's standard V2 VBR preset.... well at least in respect to the file sizes it produces. I use the same quality setting whether I'm encoding stereo or multichannel audio.

I'm pretty sure foobar2000 doesn't have an AC3 encoder preset so you need to make your own. Once in the encoder setup section you'd add a new encoder and choose, "custom", then point foobar2000 to wherever you've put the AFTEN encoder. For the command line, this works fine with foobar2000. The -b parameter being the bitrate, which you can of course change. If you don't specify one, Aften picks it's own. http://aften.sourceforge.net/longhelp.html

-readtoeof 1 -v 0 -b 448 - %d

And one you get to know foobar2000.... which admittedly can take a little time..... you'll probably realise it's a very good audio player.

Actually foobar2000 works for FLAC and after installing the DTS Decoder, It works perfect for converting DTS files too!
Thanks man
And as you said it doesn't support AC3, I'm gonna try what you said.
On the topic, What about ABR/VBR/CBR? Which one works better? Lets say I want to convert a FLAC file ( 7200 kbps, 16 bit ) to AAC ( 320 kbps ).

[sarofski]

For conversion of DTS 5.1 to AC3 5.1 eac3to works fine

I guess I'll try this one too!

[hello_hello]

On the topic, What about ABR/VBR/CBR? Which one works better? Lets say I want to convert a FLAC file ( 7200 kbps, 16 bit ) to AAC ( 320 kbps ).

To be honest I've never used anything but the quality setting for AAC. Unless there's a specific need to use something else.... well I can't think of any reason why you would unless you need a specific size for some reason or you need a constant bitrate for streaming. It'd be a bit like using a constant or average bitrate when encoding video rather than a variable bitrate. You'd be wasting bits all over the place and probably not have enough at times.

I think generally the default quality of Q0.50 is considered to be "transparent", but you can crank it up if you like. I think at maximum quality it'll average something close to 400Kb/s for a stereo track and Q.75 should give you roughly 300Kb/s on average. Well according to foobar2000 it'll be around there when encoding music but for normal video soundtracks, especially multi-channel ones, at the same quality setting the average bitrate will vary quite a lot.

The quality setting would be analogous to using x264's CRF encoding. It just uses whatever bitrate is required to achieve the specified quality at any given time. I think Average bitrate is like a mix of constant and true variable bitrate encoding, in that the bitrate varies, but the encoder is aiming for an average bitrate so it takes it's best guess as to the bitrate to use as it goes. There is a 2 pass ABR encoding method which would probably distribute the bits better while still achieving a target average bitrate, but personally I think using the quality setting is the best option.

PS. I know for MP3 encoding a lot of people use a 320Kb/s constant bitrate, but according to the golden eared folks over at hydrogenaudio, there's never been a listening test which proves it provides any better quality or more transparency than LAME's maximum quality VBR preset, which I think averages around 245Kb/s when encoding music.
http://wiki.hydrogenaudio.org/index.php?title=Recommended_LAME#Recommended_encoder_settings

[sarofski]

To be honest I've never used anything but the quality setting for AAC. Unless there's a specific need to use something else.... well I can't think of any reason why you would unless you need a specific size for some reason or you need a constant bitrate for streaming. It'd be a bit like using a constant or average bitrate when encoding video rather than a variable bitrate. You'd be wasting bits all over the place and probably not have enough at times.

I think generally the default quality of Q0.50 is considered to be "transparent", but you can crank it up if you like. I think at maximum quality it'll average something close to 400Kb/s for a stereo track and Q.75 should give you roughly 300Kb/s on average. Well according to foobar2000 it'll be around there when encoding music but for normal video soundtracks, especially multi-channel ones, at the same quality setting the average bitrate will vary quite a lot.

The quality setting would be analogous to using x264's CRF encoding. It just uses whatever bitrate is required to achieve the specified quality at any given time. I think Average bitrate is like a mix of constant and true variable bitrate encoding, in that the bitrate varies, but the encoder is aiming for an average bitrate so it takes it's best guess as to the bitrate to use as it goes. There is a 2 pass ABR encoding method which would probably distribute the bits better while still achieving a target average bitrate, but personally I think using the quality setting is the best option.

Thanks a lot for the detailed explanation

[sarofski]

http://www.dvdvideosoft.com/products/dvd/Free-Audio-Converter.htm Free Audio Converter performs conversion of almost all audio formats to a big variety of them in the output. I'm sure, the progran can convert your files too.

I have tried it before, it doesn't support "6 ch Conversion" for AAC and AC3.
But yes, it's a very good program. Actually the whole package "Free Studio Manager" is amazing.