Prob converting DTSMA 6.1 channel

[tebasuna51]

Originally Posted by JimmyBarnes

How do you arrive at this scheme?

1) Select 6 output channels
2) Click over conexión 6 -> 6 (to delete)
3) Click over input 6 and output 5 (mix BC-SL to SL)
4) Click over input 7 and output 6 (SL -> SL)
5) Click over input 5 and output 6 (mix BC-SR to SR)

[JimmyBarnes]

Originally Posted by Ghitulescu

Or you can get only the core (5.1) - this can be directly translated into DD 5.1, no channel remapping.

As indicated before, I tried:

eac3to SW3.dtsma SW3.dts -core

but it gave:
DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 1509kbps, 48kHz)

and Audacity confirmed that the DTS was 6.1, not 5.1

[Ghitulescu]

I am not expert in eac3to and curious channel mappings, I assume you should use remapping function of eac3to if the direct extraction does not work. As I said, I keep the original.

[tebasuna51]

Originally Posted by JimmyBarnes

As indicated before, I tried:

eac3to SW3.dtsma SW3.dts -core

but it gave:
DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 1509kbps, 48kHz)

and Audacity confirmed that the DTS was 6.1, not 5.1

Can you upload a sample of this dtsma?, for instance with:

eac3to SW3.dtsma Sample.dts -20mb

[JimmyBarnes]

Originally Posted by tebasuna51

1) Select 6 output channels
2) Click over conexión 6 -> 6 (to delete)
3) Click over input 6 and output 5 (mix BC-SL to SL)
4) Click over input 7 and output 6 (SL -> SL)
5) Click over input 5 and output 6 (mix BC-SR to SR)

Thanks but what I meant was, How do you determine what channels to mix?

What are the rules governing downmixing which make your scheme correct and mine wrong?

TIA

[JimmyBarnes]

Originally Posted by tebasuna51

Can you upload a sample of this dtsma?, for instance with:

eac3to SW3.dtsma Sample.dts -20mb

I created sample DTSMA, uploaded to MediaFire:

https://www.mediafire.com/?jtis55sjhxssu3g

[Ghitulescu]

Originally Posted by JimmyBarnes

What are the rules governing downmixing which make your scheme correct and mine wrong?

For the simple reason that there is a rule on how to distribute the sounds amongst channels. If one combines them back (downmixes) then he had to combine the corresponding sounds, not to pick up arbitrary channels, for other reasons.

[tebasuna51]

Originally Posted by JimmyBarnes

Thanks but what I meant was, How do you determine what channels to mix?

What are the rules governing downmixing which make your scheme correct and mine wrong?

Here you have all DTSMA channel layouts: http://forum.doom9.org/showthread.ph...33#post1714833

And here a channel test sample for all the layouts: https://www.sendspace.com/file/pd2o0f

Your sample is the I: 6.1-L,R,C,LFE,Ls,Rs,Cs discrete (I61_disc.dtshd file)

When a proper decoder (ArcSoft, dcadec, ffmpeg) convert this dtshd to PCM samples the channels order must be the same than in WAV specs:
Track 1: Front Left (L in dts)
Track 2: Front Right (R in dts)
Track 3: Front Center (C in dts)
Track 4: Low Frequency (LFE in dts)
Track 5: Back Center (Cs in dts)
Track 6: Side Left (Ls in dts)
Track 7: Side Right (Rs in dts)

Like the channel layout for PCM 5.1 is:
Track 1: Front Left (FL)
Track 2: Front Right (FR)
Track 3: Front Center (FC)
Track 4: Low Frequency (LF)
Track 5: Side Left (SL)
Track 6: Side Right (SR)

The 4 first tracks are the same but the last 2 must be:
Track 5: SL + 0.707*BC
Track 6: SR + 0.707*BC

The BC channel contribution is added to Side channels with half acustic power (-3dB in volume) in each.

With your scheme you have:
Track 4: LF + BC
than have no sense.

Originally Posted by JimmyBarnes

I created sample DTSMA, uploaded to MediaFire

I can't reproduce your problem (AC3 with more length than DTSHD) with this sample. Sorry I can't help with that.

[JimmyBarnes]

Originally Posted by tebasuna51

Here you have all DTSMA channel layouts: http://forum.doom9.org/showthread.ph...33#post1714833

Thanks, this tells me what I want to know.

cheers

[SeeMoreDigital]

Originally Posted by JimmyBarnes

I created sample DTSMA, uploaded to MediaFire:

https://www.mediafire.com/?jtis55sjhxssu3g

Well...

I've just dragged and dropped the (dts-hd ma+dts-es) sample file into UsEac3to v1.1.7 (with eac3to v3.29) and it managed to create a 640Kbps 5.1Ch AC3 stream without any issues at all

[Sparktank]

I thought his original log shows that it downmixed to 5.1 without errors.
Just some minor normalization (-0.9dB). or something.

I think it's mostly an Audacity issue after using eac3to.
Audacity can use external libraries for FFMPEG, but won't update internally (because it's too much work for them).
I tried to contact them several times about updating their FFMPEG to current stable releases (if not Zeranoe releases).

Trying to keep the 6.1 discreet is kind of useless.
The previous solutions are best: downmix to 5.1 or upmix to 7.1 (double-back from center-back).

[JimmyBarnes]

Originally Posted by SeeMoreDigital

Well...

I've just dragged and dropped the (dts-hd ma+dts-es) sample file into UsEac3to v1.1.7 (with eac3to v3.29) and it managed to create a 640Kbps 5.1Ch AC3 stream without any issues at all

Length of the DTSMA and AC3 are the same for the sample?

eac3to 3.29 (command-line) did the full conversion, but the AC3 was 14.3 % longer than the DTSMA.

[tebasuna51]

Originally Posted by JimmyBarnes

Length of the DTSMA and AC3 are the same for the sample?

Yes, your dts sample is 40.459 sec. and ac3 40.480 sec. (1265 frames, with 1264 frames is only 40.448).

Do you have the same problem with this sample?

[JimmyBarnes]

Originally Posted by tebasuna51

Yes, your dts sample is 40.459 sec. and ac3 40.480 sec. (1265 frames, with 1264 frames is only 40.448).

Do you have the same problem with this sample?

Yes, using eac3to CL, the resultant AC3 is 46.272 s, 14.37 % longer than the source DTSMA.

So, what is it that UsEac3to v1.1.7 does, it's supposed to be just a GUI for eac3to, isn't it?

[tebasuna51]

Originally Posted by JimmyBarnes

Yes, using eac3to CL, the resultant AC3 is 46.272 s, 14.37 % longer than the source DTSMA.

So, what is it that UsEac3to v1.1.7 does, it's supposed to be just a GUI for eac3to, isn't it?

I use also UsEac3to but like you say is only a GUI than create a command line like your:

eac3to v3.29
command line: C:\BDrip\EAC3to\eac3to SW3.dtsma SW3_640norm.AC3 -640 -normalize

I tested all variants using -normalize/-down6/-dcadec ... and always work fine.

Maybe is your ArcSoft version, try using -dcadec.
With ArcSoft 1.1.0.0 work fine also.

[JimmyBarnes]

Originally Posted by tebasuna51

Maybe is your ArcSoft version, try using -dcadec. With ArcSoft 1.1.0.0 work fine also.

Worked OK using libDcaDec.

I tried implementing ArcSoft 1.1.0.1 downloaded here:
http://www.dvbsupport.net/download/i...ct=view&id=277

It says: "Just put dtsdecoderdll.dll and the related .CRT folder (otherwise it will not run if the related Visual C++ Redistributable package is not installed on the system) into base folder of the programs."

I did that, however eac3to couldn't get it to work and used libDcaDec instead => AC3 of correct length

Was using tehparadox unspecified ArcSoft DTS decoder version before (which eac3to used but gave bad 6.1> 5.1 AC3), will try this one:

https://www.sendspace.com/file/ief0t0 => ArcSoft DTS decoder 1.1.0.0

YES finally, a good conversion to AC3, so looks like bad ArcSoft DTS decoder was the problem..

[tebasuna51]

Actually ArcSoft is only necesary to decode DTS-Express, in all my test libDcaDec decode bitidentical outputs than wav sources (ArcSoft 1.1.0.0 seems have, in few cases, a rounding error and the output differ than sources in the last significant bit).

[JimmyBarnes]

Originally Posted by tebasuna51

Actually ArcSoft is only necesary to decode DTS-Express, in all my test libDcaDec decode bitidentical outputs than wav sources (ArcSoft 1.1.0.0 seems have, in few cases, a rounding error and the output differ than sources in the last significant bit).

That's interesting, I always thought ArcSoft was supposed to be the best tho I have not done any testing.

[tormento]

Originally Posted by tebasuna51

Your downmix is wrong, you mix the Back Center channel to LFE channel.

Sorry to resurrect a very old thread but what is the correct way to upmix 6.1 to 7.1 with ffmpeg?

[tebasuna51]

If your 6.1 is a dts matrixed (BC analog mixed in SL-SR) the recommended way is let it as 5.1

If your 6.1 is discrete (independent BC channel) the recommended way is downmix to 5.1 to obtain the same 2D surround, with less bitrate needed, using:

ffmpeg -hide_banner -i INPUT_61 -vn -filter_complex^
 "asplit [f][s]; [f] pan=3.1|c0=c0|c1=c1|c2=c2|c3=c3 [r]; [s] pan=stereo|c0=0.35*c4+0.5*c5|c1=0.35*c4+0.5*c6, compand=attacks=0:decays=0:points=-90/-84|-8/-2|-6/-1|-0/-0.1, aformat=channel_layouts=stereo [d]; [r][d] amerge [a]"^
 -map "[a]" OUTPUT_51

Where OUTPUT_51 can be AAC, OPUS or EAC3

If you need 2D surround 7.1 (I don't know for what) you can use:

ffmpeg -hide_banner -i INPUT_61 -vn -af pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=0.707*c4|c5=0.707*c4|c6=c5|c7=c6 OUTPUT_71.wav

And recode your 7.1 2D