DTS to AC3 problems

[ckamc]

I have been frustrated with this for a while now.

I found quite a few guides, both foobar, besweet, eac3to, HypercubeTranscoder... and I can't seem to get any of them to work.

foobar produces a wav file but I can automatically tell there is a quality loss (lots of noise),
besweet would not want to encode,
eac3to wouldnt want to encode either,
HypercubeTranscoder would get stuck at 33% of the way thru (aka just stop)

I noticed that a few of these guides are dated, and I was wondering if there is any new methods or easier ways to go about this conversion.

mkv2vob conversion of DTS produces a lot of noise as well (mainly during the dramatic scene's just like foobar)

I am trying graphedit but can't find an file needed for the process (DTSWAVSource.ax)

[tebasuna51]

I found quite a few guides, both foobar, besweet, eac3to, HypercubeTranscoder... and I can't seem to get any of them to work.

BeSweet and Hypercube are outdated, but eac3to and foobar (with the foo_input_dts.dll) can't have problems with standard DTS.
Yo can try also BeHappy or other AviSynth methods.

[ckamc]

BeSweet and Hypercube are outdated, but eac3to and foobar (with the foo_input_dts.dll) can't have problems with standard DTS.
Yo can try also BeHappy or other AviSynth methods.

With the BeHappy method I found the nicaudio.dll but an error of Error: Can't find audio stream! comes up.

I look to find that "NicDTSSource" function is not found during preview...

any suggestions?

[tebasuna51]

With the BeHappy method I found the nicaudio.dll but an error of Error: Can't find audio stream! comes up.

I look to find that "NicDTSSource" function is not found during preview...

any suggestions?

Then you have a DTS-HD or corrupt file.

Try with DelayCut to fix the dts.

[ckamc]

while I dig up Delaycut

I went back and tried foobar again, using aften

I checked with a transcode guide and found that my parameters were incorrect

so I changed them to

-v 0 -b 448 -fba 0 -fes 0 %s -0 %d

it encodes, but then fails to playback the created file and which when I look for it is no longer in existance :rolleyes:

[ckamc]

BeHappy says the same thing

could it be where I have the nicaudio dll saved? I pretty much put one in the parent directory and each one of the source directories since the guide I found did not specify where exactly

trying foobar again to copy the message that it left me last time, if it comes up with this dts fix.

the bitrate is 1535 which I have been told is not HD

[ckamc]

Here is the message from foobar

An error occured while finalizing the encoding process (Could not start commandline encoder) : "G:\Track2_fixed.ac3"

produced a file in temp of over 2.2gb while the original is 1.2

[ckamc]

Well tried wav which produced a horrible quality version before, but some how this time around it did things correctly.

Now I just hope using dkaudio for wav to ac3 won't hinder the quality

wav file came out 3.8GB

[roozhou]

try decoding with ffmpeg

[tebasuna51]

try decoding with ffmpeg

The libav used in eac3to is the same than used in ffmpeg with some improvements (channel mapping, DialNorm, DRC)

@ckamc
Put the logs of eac3to and DelayCut or upload a fragment of your dts.
We can't help you without some info.

[boykster]

if it's a straight dts file (not HD) you should be able to


eac3to inputfile.dts outputfile.ac3 -libav -640


that's how I convert DTS to AC3 (just did it last night on a movie) and it works fine. If it is a DTS-HD file, I would first run:


eac3to inputfile.dts outputfile.dts -core


to extract the core DTS stream, but it sounds like you have a dts stream to begin with.

[eas4uk]

Alternatively you can also use tranzcode on the dts file - this will then generate a multichannel wav file which you can feed to aften to get your ac3 file

[lariva]

You have probably checked before but - make sure your sampling rate and width are aligned. DTS can come with 14 bit 16bit and 24 bit.... along with 41 and 48khz
For example pink floyd dts file that i have extracted from a dvd is:

4:24.082 (11646000 samples)
44100 Hz
6 channels
1411 kbps
DTS 3/2.1 (5.1) 14bit LE

Whereas anther DTS track from another DVD is
5:35.276 (16093257 samples)
48000 Hz
6 channels
755 kbps
DTS 3/2.1 (5.1) 16bit BE

Whenever you are transcoding to another format, make sure you take that into an account. Also in foobar there is preferred depth output setting which seems to be getting involved when you wav convert.

In short - pay attention to sampling / depth values - don't assume automatic conversion will take place (i.e. SPDIF output in foobar and 41 / 48khz situation)

[tebasuna51]

You have probably checked before but - make sure your sampling rate and width are aligned. DTS can come with 14 bit 16bit and 24 bit.... along with 41 and 48khz
For example pink floyd dts file that i have extracted from a dvd is:

4:24.082 (11646000 samples)
44100 Hz
6 channels
1411 kbps
DTS 3/2.1 (5.1) 14bit LE

Whereas anther DTS track from another DVD is
5:35.276 (16093257 samples)
48000 Hz
6 channels
755 kbps
DTS 3/2.1 (5.1) 16bit BE

Whenever you are transcoding to another format, make sure you take that into an account. Also in foobar there is preferred depth output setting which seems to be getting involved when you wav convert.

In short - pay attention to sampling / depth values - don't assume automatic conversion will take place (i.e. SPDIF output in foobar and 41 / 48khz situation)

Ok with samplerate comments, but you are wrong about the bitdepth.

Don't mistake the dtswav format (14bit LE) with 14 valid bits from 16, and the 2 more significative set to 0 to protect the speakers if played as wav, with sample bitdepth like in wav files.

All dts can be decoded to 24 bitdepth.