How do you convert DTS to PCM

[ChrisDaMan]

turns out the ps3 can't read DTS off the hard drive, so in order to maintain my quality, i want to take my DTS and decode it on the computer to PCM which my ps3 will then be able to read. Now i have tried doing this with MediaCoder, but it seems to take my 24 bit audio and scale it down the 16 bit. it also ends up sounding funny, i want to keep the 24 bit audio. I have considered using EAC3to, but every time i try and open the program, command prompt opens runs a script or something, closes, and then nothing happens, so, idk if its cuz it won't run on win7 or something, but it doesn't work.

so, anyone know how to transcode my 24bit DTS into a 24 bit PCM

[poisondeathray]

eac3to is a command line application . open up a command prompt and type

eac3to input.dts output.wav


or you can try one of the gui's for it (they're actually harder to use, you have to setup paths and configurations before you use it)

[ChrisDaMan]

eac3to is a command line application . open up a command prompt and type

eac3to input.dts output.wav


or you can try one of the gui's for it (they're actually harder to use, you have to setup paths and configurations before you use it)

yeah, the GUI defiantly does seem more complicated, but, i sure don't seem to understand what it is that ur saying, go into more details please... and strait up opening up command prompt and typing in eac3to it says that that command doesn't exist.

[poisondeathray]

you have to navigate to the folder using "cd" for "change directory"

so if current folder was c:\documents and settings\user\

and the level you want to go to was c:\documents and settings\user\somefolder

just type "cd somefolder"

it's easier if the files are in the same folder (all the audio files as the eac3to folder)



maybe some other programs easier; try foobar2000

[ChrisDaMan]

ok, im using the command line, but now that i got it open and all, how do i convert my .dts file to a .wav? what would i type in, and yes i put my DTS file in the eac3to folder

[poisondeathray]

eac3to input.dts output.wav

[ChrisDaMan]

thank you very much, do you know where i could download arcsoft DTS decoder?

[poisondeathray]

it comes with total media theatre or related products (not free)

is this vanilla 5.1 DTS ? the libav decoder should work fine

you can force a decoder

e.g.

eac3to input.dts output.wav -libav


here are some reference pages / guides
http://forum.doom9.org/showthread.php?t=125966
http://en.wikibooks.org/wiki/Eac3to/How_to_Use

[ChrisDaMan]

now i have a problem... my wav file seems to be fine when looking at the files size, took my 1 GB dts and made it almost 5GB, only problem is... when i play it back, it says its something like 12 min long.. WTF, and when i try and mux it with tsMuxer, it freezes after showing LPCM bad frame detect resync stream several times.

[poisondeathray]

what kind of dts ?

what does the eac3to log file say ?

can you play the wav in winamp or mpc ?

[poisondeathray]

I forgot the header isn't compatible with tsmuxer

use pcm2tsmu

see
https://forum.videohelp.com/threads/305153-DTS-HD-to-LPCM-and-PS3-How-I-Did-It

you might need lpcm instead of wav (differences in header)

[ChrisDaMan]

ok, thanx its working now

[ian curtis]

im late to the game here

I have a similar problem (cant figure eac3to for the life of me ,command line)

So i tried foobar2000 . i convert DTS to flac at best smallest compression
8 , the only problem is it triples the size ,3 to 9 gigs , which i cant burn to
and dvd disc .

Keeping the high 24 bit quality ,is there a method to make the resulting pcm
MUCH smaller ?

Thanks

[hello_hello]

Not losslessly. 24 bit lossless multi-channel files will be huge. I thought eac3to decodes DTS to 16 bit by default? Maybe not.....

If it's just the DTS core then it doesn't actually have a fixed bitdepth so you can decode it to whatever bitdepth you want. I'd assume as DTS-HD is lossless it'd have a fixed bitdepth..... but I don't really know how that works, given it's a combination of a lossy core plus the additional HD part to make it lossless.

But anyway..... there's no magic way to have lossless 24 bit audio in small files sizes I'm aware of. If you want to keep the audio in a lossless format, 16 bit is probably the only realistic option.

[Island_Dweller]

I'd assume as DTS-HD is lossless it'd have a fixed bitdepth..... but I don't really know how that works, given it's a combination of a lossy core plus the additional HD part to make it lossless.

Yes, it's lossless, on BD, it will be 16 bit or 24 bit, depending on what the studio decided to include in the BD.

[El Heggunte]

DTS-HD Master Audio is lossless. "Pure" DTS-HD is lossy.

Besides, none of the versions of Arcsoft's DTSdecoderDLL.dll on which eac3to relies is "perfect" --- they do not decompress 2.0 DTS-HD Master Audio losslessly, for example.

[ian curtis]

Thanks mates

What still confuses me is , i have lossless 24 bit files ( even wave ) , 24 /192 ,2 qnd multi channel that are a fraction the size
of what i get when i convert this DTS to 24 bit flac

What is it about it being a DTS from the go that makes it convert so large ,when no other 24/192 2 or multi channel flac
ive eve seen is that large ?

Cheers mates

[El Heggunte]

^ Please post a Mediainfo report about one of those DTS files
( you may omit its filename though, since we don't need such kind of information ).

[ian curtis]

Thank you

Do you need screens of all the sheets ? Or just one in particular ?

Thanks

[ian curtis]

PS

i have to be honest here , im daft

im mucking around with all the views and am not seeing any info i dont
already know , Ie file name/size/iso etc

Any ideas what to do once i load the iso ia appreciated

Cheers

[ian curtis]

Heres a foobar screen of properties , i hope it helps

i get nowt from mediainfo because im daft

Cheers

PS i cant upload attachments of the screen for some reason UGH

[ian curtis]

ill try again

[hello_hello]

Thanks mates

What still confuses me is , i have lossless 24 bit files ( even wave ) , 24 /192 ,2 qnd multi channel that are a fraction the size
of what i get when i convert this DTS to 24 bit flac

What is it about it being a DTS from the go that makes it convert so large ,when no other 24/192 2 or multi channel flac
ive eve seen is that large ?

Are you sure those smaller files really are 24 bit?
I'm pretty sure if you convert lossless 16 bit audio to flac (for example) while specifying a bit depth of 24, you'll still only have 16 bit audio but it'll be in a 24 bit container.... for want of a better way to explain it.... because I don't fully understand how it all works.

One way to test if the 24 bit files really are 24 bit, might be to re-encode one as flac again while specifying 16 as the bit depth. If you go from flac to flac and the file size doesn't change much then I'd assume the original only contains 16 bit audio, even though to foobar2000/MediaInfo etc it's a 24 bit flac file.

You could probably also test it out by converting a 16 bit wave file to flac while specifying 24 bit. I'm pretty sure you'll still have 16 bit audio in the flac file. Foobar2000 doesn't seem to mess with the bitdepth unnecessarily.
If you take the same audio and encode it using a lossy encoder, then convert the lossy version to flac while specifying 24 bit, you'll probably end up with a much larger flac file the second time, because as you're going from a format with no bitdepth to a format which has one, foobar2000 should then give you whatever bit-depth you specify.

[ian curtis]

Thanks mate

ill give that a wing tomorrow when my eyes wont be on fire lol

But in mean time can i tell you inside each iso is one BDMV folder
with one single M2TS file

I feel that may tell you something ,it doesnt me cuz im too daft

But can i guess its Bluray audio ?

I got more info as follows

Audio Codec : DST 2.0 5.1 2,82 mHz Rip type : Image.iso Bitrate : lossless source
All Tracks are in one SACD Stereo and a triple in Multichannel SACD: SACD 1 in 5.1

Its a 4 disc set , they all give me the same over size results when i convert to
pcm flac


Cheers

[hello_hello]

I just tried a few MP3 to flac encodes with foobar2000. With the bitdepth setting on auto foobar2000 gave me a 16 bit flac file which was 29.4MB. When I specified a 24 bit output, the flac file was 51.9MB, so it makes quite a difference.

I converted the MP3 to 16 bit wave, then converted the wave file to 24 bit flac. This time the 24 bit flac file was only 29.7MB, so obviously it can't contain 24 bit audio even though it's seen as a 24 bit flac file.

I had a thought..... Aside from converting the MP3 file to wave, the above encodes didn't use dithering , so I tried the wave file to 24 bit flac encode again while applying dithering, and that caused foobar2000 to change the audio bitdepth from 16 to 24 bit. At least I assume it did, the flac file size was 53.8MB.
Not that you'd be likely to have a reason for increasing the bitdepth when converting from one lossless format to another, and I can't imagine why there'd often be a need for dithering either, but I thought I'd mention the possible effect of dithering on the output audio bitdepth.

Which does make me wonder what happens if you use a DSP while encoding a 16 bit lossless audio file and specifying 24 bit flac as the output. My little test encodes didn't use any DSPs. I'll have to test it later..... out of curiosity.

[ian curtis]

Hi mate

Thanks for a reply

im shy to say i m so new i have no idea what a DSP is , i know a DSD is an SACD tho,probably un related ? And dithering etc UGH !!!!

heres some technical info i gained from the files ,you sound smart ,so this may make sense to you mate

Audio Codec : DST 2.0 5.1 2,82 mHz Rip type : Image.iso Bitrate : lossless source
All Tracks are in one SACD Stereo and a triple in Multichannel SACD: SACD 1 in 5.1
In foobar info
subsong index: 2
file size: 3.26 GB
Duration: 5:.50.480 (9989 194 752 SAMPLES)
Sample rate:2228400Hz
Channels:6
Bits per sample: 24
Bitrate: 16394 kbps
CodecTS64
Encoding" lossless

[hello_hello]

I don't know much about DSD or SACD, or for that matter even understand how the sampling method works, but it's not 16bit or 24bit like PCM audio, it uses a 1bit sampling method, but as I said, I don't really understand it, so I don't understand the way it's labelled or where DTS comes into it. http://en.wikipedia.org/wiki/Direct_Stream_Digital
If it's actually a DSD disc image maybe foobar2000 is displaying how it'll be decoded as it's got to be converted to PCM for playback using a PC. http://en.wikipedia.org/wiki/Direct_Stream_Digital#DSD_disc_format

DSP = Digital Signal Processor. Foobar2000 comes with several of them or you can download DPS components. They'll be included in the components list and can be applied when setting up the converter (limiter, 5.1ch mixdown etc).

Dithering is a form of noise introduced into the original audio signal. Sources which use a fixed bitdepth (16 bit or 24 bit etc) can only apply a fixed number of values to each sample. Obviously the greater the bitdepth the greater the accuracy. When converting from a format which has no fixed bitdepth (analogue or lossy codecs etc) to one which does, or converting a higher bitdepth source to a lower one, the difference between the original and the output is caused by the "rounding" to fixed values and is referred to as quantization error. You'd probably need to convert to a fairly low bitdepth for it to be obvious but the result in terms of what you might hear is referred to as quantization noise. The theory behind introducing a small amount of randomness into the original signal is to reduce the quantization noise.

When you select an encoder which uses a fixed bitdepth in foobar2000's converter setup, the dropdown boxes appear below for bitdepth and dithering. I'm pretty sure "auto" for bitdepth gets foobar2000 to convert lossy sources to 16 bit while it leaves other sources at their original bitdepth. I originally thought it might convert sources with bitdepths greater than 16 to 16 bit but I now suspect that's not correct. The dithering options are "always", "never", and "lossy sources only". Dithering can be used in all sorts of digital processing, not just for audio.

So what seems to happen is if you take a 16 bit source and convert it (no extra processing) to another lossless format, foobar2000 will still give you a 16 bit output with the bitdepth setting on auto, and also if you select a higher bitdepth.... which makes sense given increasing the bitdepth wouldn't make the output any more accurate than the original. At least that's how it works for flac. So if you convert a 16 bit wave file to a 24 bit flac file, the audio within will still only be 16 bit even though the flac file itself is theoretically 24 bit, so it'll be much smaller than an equivalent 24 bit flac file which actually contains 24 bit audio.
Once you apply any sort of processing to the source when converting however, as it modifies the original audio, foobar2000 will output a higher bitdepth if you tell it to, which also makes sense as theoretically it'd be more accurate. Well I've tested it while applying dithering and it certainly gave me a true 24 bit flac file from a 16 bit wave file, but I've not testing it when using any of the DSPs. Logically though, it'd work the same way. In fact I'd be fairly surprised if it didn't, as I'm pretty sure all foobar2000's internal audio processing calculations use 32 bit floating point for a high degree of accuracy.

[Island_Dweller]

"I'm pretty sure all foobar2000's internal audio processing calculations use 32 bit floating point for a high degree of accuracy."

Most probably, seeing as something as simple as changing the volume is a floating point operation.