eac3to - audio conversion tool

[Music Fan]

ffmpeg could maybe decode it.

[hightime]

Just discovered eac3to (brilliant!) in the context of decode/converting TrueHD. The 7.1 audio I have needs to be converted to 5.1 so I used -down6 but I am curious about how it works. The side channels must get mixed into the back channels but do they also get mixed into the front channels? Also, how is clipping avoided? I converted the audio to a .flac file but this would not load into audacity so I made another conversion to .wav but this appeared to only convert about half the length of the audio. Should I use .pcm instead, or something else? The converted .wav file length is long (over 7 gbytes).

Apologies if these topics have been covered (I searched but couldn't find anything ... and I'm quite new to this ... ).

with thanks.

[LigH]

WAV is restricted to 32 bit chunk sizes (<4 GB; depending on the interpretation of the first bit as sign, in carelessly programmed tools, possibly even <2 GB). You can try ".wavs" which generates several mono WAV files per channel, location appended to filename. Or try ".w64" for the WAV64 format.

[kit90]

When eac3to downconverts a 24-bit audio file to 16-bit, it applies Triangular PDF dithering automatically. Does it also apply any noise-shaping?

[madshi]

No noise shaping, only TPDF dithering.

[kit90]

No noise shaping, only TPDF dithering.
Thanks :)

[ACrowley]

Hi

I have a Problem with some AC3 5.1 Tracks from a HDTV TS Capture.
Its surely 5.1 Channel, but eac3to detects it as 2.0 and outputs only 2 Channel Left and Right .wavs

Is there a Way to correct the Header Info in the AC3 or a override in eac3to to Output 6 .wavs ?

thanks

[Music Fan]

This is maybe 2.0 followed by 5.1 in the same track, it happens on some tv channels (movie in 5.1, ads in 2.0).
TSdoctor can cut audio/video on audio format change but it's not free.

[ACrowley]

Hi

I cut the TS Streams with Videoredo Frame Accurate

And i tried TS Doctor with the AC3 Function, but eac3to still detects it as 2.0 AC3

EDIT
When i enable the TSDoctor Option Insert Ac3 5.1 Frames if needed then eac3to detects 5.1 AC3 and decodes to 6 CH .wavs

But the demuxed AC3 from this fixed TS Stream shows still 2.0 again ;)

What works: i simply cut a few Single Frames at the Stream Beginning with Videoredo, then its 5.1 Channels :)

[Q-the-STORM]

you can use ProjectX for that...
Go to PreSettings -> Audio -> check "replace all non-3/2 AC-3 by 3/2lfe silence"

demux the ac3, open it with projectX and click on "Quick Start"...
ProjectX will then replace all 2.0 audio in the track with 5.1 silence, making it possible for eac3to to detect it as 5.1 audio...

a feature like that was actually requested on the VideoReDo forums years ago, but it was never implemented...

[Yoshi]

I might have found another bug - this time in the decoder used for Dolby TrueHD tracks in conjunction with Dolby Atmos encodings (eac3to 3.31).

In the case of Léon (US Supreme Cinema Series), the decoding of the Atmos track leads to weird cracklings at certain positions whereas the decoding of the embedded AC3 track is flawless.

http://www.bilder-upload.eu/thumb/399a48-1451222081.jpg (http://www.bilder-upload.eu/show.php?file=399a48-1451222081.jpg)

[Boulder]

Have you tested decoding with a recent ffmpeg build? If it also causes the issue, you need to report the bug to the ffmpeg devs.

[Yoshi]

Have now. Using ffmpeg-20151227-git-baf4c48-win64-static to decode the thd file demuxed by eac3to, it's the same result.

http://www.bilder-upload.eu/thumb/bcb20c-1451257105.jpg (http://www.bilder-upload.eu/show.php?file=bcb20c-1451257105.jpg)

Guess I better let them know.

Still confused - wasn't the TrueHD decoder to be bug-free so far at least?

[nevcairiel]

There is no guarantee your TrueHD stream isn't just broken.

[Yoshi]

@nevcairiel

Which is why I wanted to take some precaution and used the word "might" wisely. Guilty as charged that the source is not really an official one but to double-check this, I used a remuxed one and a nominal 1:1 copy which has to be put together by eac3to.

Just for my defense - NO, I won't buy that movie for the fourth time (two DVDs and the LaserDisc is enough now), just because they finally get a half-way decent video master they could have come up with in the first place (so annoying).

Besides that, I don't consider this Atmos remix to be that great due to its "tinny" acoustics and of course, the 4K remastered Blu-ray doesn't contain the 5.1 mix.

FFmpeg states something about "lossless" check failed, but only with one of the two sources as far as I remember - the crackled PCM result is similar though.

Maybe someone can grab the original and double- (or rather: triple-) check.

[Yoshi]

Another oddity:

When decoding this DTS-HD MA track, there is some kind of contradiction in the log output of eac3to: on one hand, libDcaDec is allegedly outputting 16 bit data, however the resulting PCM file uses all the 24 bits according to eac3to at the same time.

In any case, the output at least matches the ArcSoft decoder's. Yet again, I don't know if it's really lossless or not.

http://www.bilder-upload.eu/thumb/ff5e1b-1451441001.jpg (http://www.bilder-upload.eu/show.php?file=ff5e1b-1451441001.jpg)

[ACrowley]

Have anbody here succes with the Sonic Audio Decoder on W7/8.1 x64 ?

I tried v4.2 and 4.3 from the Decoder Pack, also v5 from, Sonic Cinevision.
The Sonic Cinemaster® Audio Decoder 4.3 appears in Directshow registered Filters and i can use it MPC HC etc

But eac3to gives me :

The Sonic Audio Decoder (3.31.0.0) doesn't seem to be installed

I placed the Files in Systems Folder etc etc, nothing helps.
Its a clean System without any Decoder Packages etc.

Is there trick on W8 x64 ? :)

Thanks

[dts350z]

I would like to resample to 32 bit float, and thus avoid any 2nd pass. Can eac3to do that?

ideally output would be in a format that also supports files larger than 4GB, such as w64 or rf64.

[dts350z]

Anybody notice that the actual usage output of the current version doesn't show the:

-r8brain use r8brain resampler instead of SSRC

option, unlike the output shown in the post at the top of this thread?

Version number match :confused:

[Nico8583]

Hi :) I would like to know : what is the best way to convert DTS 5.1 to AC3 5.1 ? Only "eac3to.exe source.dts destination.ac3 -640" ? Or "-normalize" or "-dontPatchDts" are need ? Thank you !

[Overdrive80]

@madshi

Instead of using r8brain, would not it be better use ffmpeg (soxr)?

http://src.infinitewave.ca/

[dts350z]

Hi :) I would like to know : what is the best way to convert DTS 5.1 to AC3 5.1 ? Only "eac3to.exe source.dts destination.ac3 -640" ? Or "-normalize" or "-dontPatchDts" are need ? Thank you !


That's a lossy to lossy conversion. Why would you want to do that?

IMHO dts is a better sounding format anyway.

[mariner]

@kukushka
Maybe we can open a new thread to speak about AAC and muxer/demuxer's (Mp4Muxer/Mp4Box/MkvToolnix/tsMuxeR), but the relevant questions for this thread are clear:

- The directshow Nero 7 decoder, used by eac3to to decode .aac, cut the first 1024 samples (21,333 ms in 48 KHz) in 2.0 and is broken for 5.1.
Take in mind the problem.

- The encoder NeroAacEnc.exe, user by eac3to to encode .m4a, put the correct delay to compensate and can be decoded without problems with NeroAacDec, Qaac or ffmpeg. No problem with it.

- eac3to works fine extracting AAC from MKV/TS/M2TS containers, I obtain the same aac than I muxed previously.
Then, to avoid problems, you can use eac3to to extract and Qaac or ffmpeg to decode (or LWLibavAudioSource inside AviSynth).

Greetings tebasuna51.

Thanks for the post. I'm using eac3to to convert aac to ac3, and struggling to understanding the delay issue. Appreciate if you could assist.

1. If direct from aac to ac3, 5ms silence seems to be inserted. This appears to agree with ffmpeg. Audacity is used for comparing the two.

2. If going from ts/mkv to ac3, Nero Audio Decoder would be used. This would seem to remove 42ms if the audio begins with silence. So the resulting ac3 would be either -37ms shorter or +5ms longer, depending on the initial content.

Does this behavior look correct to you? Is there a way not to use Nero for step 2?

Many thanks and best regards.

[tebasuna51]

Hi :) I would like to know : what is the best way to convert DTS 5.1 to AC3 5.1 ? Only "eac3to.exe source.dts destination.ac3 -640" ?
Is not the best.
Is correct using eac3to (dcadec) and Aften encoder.

But, talking about free soft, now is better ffmpeg:
ffmpeg -acodec libdcadec -i source.dts -acodec ac3 -center_mixlev 0.707 -surround_mixlev 0.707 -ab 640k destination.ac3

Or "-normalize" or "-dontPatchDts" are need?
Nope.

That's a lossy to lossy conversion. Why would you want to do that?

To save space and/or make compatible with some players, I supose.

IMHO dts is a better sounding format anyway.
Better format? I don't think so.

Talking about size/quality AC3 is better and AAC much better.
Talking about compatibility AC3 is better.

[LigH]

The "core" dts audio format on DVD Video may be a low-loss format when using the bitrate close to LPCM 16-bit stereo (1536 kbps); but it is not lossless. And the bitrate close to LPCM 16-bit mono (768 kbps) is even audibly lossy. It may use less psycho-acoustic filtering than AC3. But that doesn't make it "better": dts tries to retain even probably inaudible frequencies only "audiophiles" believe to recognize (but couldn't prove), so it possibly lacks in accuracy for audible frequencies instead.

Its purpose was to compress multi-channel audio to bitrates close to comparable usual bitrates of 16-bit LPCM in mono or stereo, and it is still a format based on 16-bit integer samples. Dolby Digital (AC3) instead works with floating point parameters, therefore it may have a better dynamic range if it compresses original 24-bit integer or even floating point samples (important especially for almost slient scenes, like in classical music), keeping the audio audible despite a loss of frequency parts excluded by psycho-acoustic filters, because you will probably not recognize them anyway (except on a rather psychological level).

[tebasuna51]

1. If direct from aac to ac3, 5ms silence seems to be inserted. This appears to agree with ffmpeg. Audacity is used for comparing the two.
This is the default behaviour for ALL ac3 encoders.
BTW, with Aften, you can avoid the insertion of 5 ms of silence with, for instance:

eac3to INPUT stdout.wav | Aften -b 192 -pad 0 -readtoeof 1 - OUTPUT.ac3

2. If going from ts/mkv to ac3, Nero Audio Decoder would be used. This would seem to remove 42ms if the audio begins with silence. So the resulting ac3 would be either -37ms shorter or +5ms longer, depending on the initial content.
Seems you have misunderstand my post. I say:
"The directshow Nero 7 decoder, used by eac3to to decode .aac, cut the first 1024 samples (21,333 ms in 48 KHz) in 2.0"

Then, always cut 21 ms, and (without -pad 0) finish with 16 ms shorter.

The Nero Audio Decoder (NeroAacDec.exe) is not used by eac3to at all.

Is there a way not to use Nero for step 2?

eac3to only can decode aac with the directshow Nero 7 decoder.

[Nico8583]

That's a lossy to lossy conversion. Why would you want to do that?

IMHO dts is a better sounding format anyway.

I don't know if I'll convert DTS to AC3 but I feel sound variation are more important on a DTS track than AC3 track. When actors are speaking, the sound is low but when there is action the sound is high so I'm playing with my remote several times ;) but perhaps it's only a feeling or it's not related to DTS.
And also for a compatibility a little bit :D

Is not the best.
Is correct using eac3to (dcadec) and Aften encoder.

But, talking about free soft, now is better ffmpeg:
ffmpeg -acodec libdcadec -i source.dts -acodec ac3 -center_mixlev 0.707 -surround_mixlev 0.707 -ab 640k destination.ac3

So ArcSoft is useless now to decode DTS ? Do you have a conversion commandline sample with dcadec and Aften ?

Nope.
Ok :)

To save space and/or make compatible with some players, I supose.
Yes for compatibility and also for the other reason (see before)


Better format? I don't think so.

Talking about size/quality AC3 is better and AAC much better.
Talking about compatibility AC3 is better.
I don't have test AAC so I don't know about the compatibility but I could test it.

Last question, what about remapping channel ? Is there a risk to have an issue with wrong channel remapping if I convert DTS to AC3 ?

Thank you !

[AlexKane]

I don't know if I'll convert DTS to AC3 but I feel sound variation are more important on a DTS track than AC3 track. When actors are speaking, the sound is low but when there is action the sound is high so I'm playing with my remote several times ;) but perhaps it's only a feeling or it's not related to DTS.
And also for a compatibility a little bit :D


The dynamic range of the source material (the sound variation you describe above) is intentional. Films are mixed for theaters, not small apartments and tiny TV speakers. Also, i believe DTS, AAC, Vorbis, Opus, etc maintain the dynamic range of the source material, since they don't apply dynamic range compression (DRC).

In the case of AC3, you can use ffmpeg to generate level scaling metadata, but the results are not ideal since obvious pumping artifacts can be introduced during playback.

[tebasuna51]

...So ArcSoft is useless now to decode DTS ? Do you have a conversion commandline sample with dcadec and Aften ?

Now dcadec is the default decoder for DTS, your sintax is enough:
"eac3to.exe source.dts destination.ac3" (640 Kb/s is the default also)

Last question, what about remapping channel ? Is there a risk to have an issue with wrong channel remapping if I convert DTS to AC3 ?
No problem.

[Nico8583]

The dynamic range of the source material (the sound variation you describe above) is intentional. Films are mixed for theaters, not small apartments and tiny TV speakers. Also, i believe DTS, AAC, Vorbis, Opus, etc maintain the dynamic range of the source material, since they don't apply dynamic range compression (DRC).

In the case of AC3, you can use ffmpeg to generate level scaling metadata, but the results are not ideal since obvious pumping artifacts can be introduced during playback.
Ok thanks, I believed DTS was the only one to use dynamic range...

Now dcadec is the default decoder for DTS, your sintax is enough:
"eac3to.exe source.dts destination.ac3" (640 Kb/s is the default also
Ok thanks :) but I don't understand why do you say "talking about free soft" for ffmpeg, dcadec and Aften are commercial softwares ?

[mariner]

Thanks for the kind reply, tebasuna51.

1.
This is the default behaviour for ALL ac3 encoders.
BTW, with Aften, you can avoid the insertion of 5 ms of silence with, for instance:

eac3to INPUT stdout.wav | Aften -b 192 -pad 0 -readtoeof 1 - OUTPUT.ac3

So, this would work for IN.aac -> OUT.ac3?
eac3to IN.aac stdout.wav | Aften -b 192 -pad 0 -readtoeof 1 - OUT.ac3

Would you also kindly provide a CLI for IN.mkv -> OUT.ac3, perhaps using ffmpeg if not possible with eac3to?

2.
"The directshow Nero 7 decoder, used by eac3to to decode .aac, cut the first 1024 samples (21,333 ms in 48 KHz) in 2.0"
Then, always cut 21 ms, and (without -pad 0) finish with 16 ms shorter.

As I'd explained, Audacity here would indicate 42ms cut, not 21ms. And only if there's silence at the beginning. Otherwise, everything is preserved by the Nero DS decoder.

So if the following command is run, the output delay is either -37ms or +5ms depending on the content.

What might be causing such unusual behavior?

eac3to IN.mkv 2: OUT.ac3

3.
eac3to only can decode aac with the directshow Nero 7 decoder.
That's interesting, given Eac3to uses libAften for decoding aac when converting aac to ac3.
Any reason?

Many thanks and best regards.

[tebasuna51]

...but I don't understand why do you say "talking about free soft" for ffmpeg, dcadec and Aften are commercial softwares ?
The DTS decoder dcadec (inside ffmpeg and eac3to) is also free soft.

The AC3 encoder inside ffmpeg is better than Aften encoder, both free soft, but maybe there are better certified Dolby Digital commercial encoders.

I say "maybe" because I don't know test about that.

[tebasuna51]

1.1
So, this would work for IN.aac -> OUT.ac3?
eac3to IN.aac stdout.wav | Aften -b 192 -pad 0 -readtoeof 1 - OUT.ac3
3.
That's interesting, given Eac3to uses libAften for decoding aac when converting aac to ac3.
Don't mistake decoder and encoder.
libAften is only a AC3 encoder, can't decode aac, that only work if you have installed the directshow Nero 7 decoder.

Recommended command line:
eac3to IN.aac -192 OUT.ac3
or
eac3to IN.mkv 2: -192 OUT.ac3

Both produce an AC3 with firts 21 ms cutted and 5 ms delayed (16 ms shorter).
You can add the parameter +16ms to both command lines to obtain the first 21 ms replaced by silence.
1.2
Would you also kindly provide a CLI for IN.mkv -> OUT.ac3, perhaps using ffmpeg if not possible with eac3to?
ffmpeg -i IN.aac -c:a ac3 -b:a 192k OUT.ac3
ffmpeg -i IN.mkv -map 0:1 -c:a ac3 -b:a 192k OUT.ac3
ffmpeg -i IN.mp4 -map 0:1 -c:a ac3 -b:a 192k OUT.ac3

All AC3 with the standard ac3 5 ms delay of silence.

2. As I'd explained, Audacity here would indicate 42ms cut, not 21ms. And only if there's silence at the beginning. Otherwise, everything is preserved by the Nero DS decoder.

So if the following command is run, the output delay is either -37ms or +5ms depending on the content.

What might be causing such unusual behavior?
Audacity don't use the Nero DS decoder to decode AAC, it use ffmpeg.

And in all my test opening IN.aac or IN.mkv or IN.mp4 the decoded aac is perfect, without any cut or delay.

I don't know how you obtain these data. Please explain your workflow.

[Thunderbolt8]

madshi, could we please get an update for dcadec? it has reached v0.2. I know we could do it ourselves but as others have reported for some reason the .dll we produce seems to work slower than yours.

[mariner]

Thanks for the kind reply,

1.


Recommended command line:
eac3to IN.aac -192 OUT.ac3
or
eac3to IN.mkv 2: -192 OUT.ac3

Both produce an AC3 with firts 21 ms cutted and 5 ms delayed (16 ms shorter).
You can add the parameter +16ms to both command lines to obtain the first 21 ms replaced by silence.

ffmpeg -i IN.aac -c:a ac3 -b:a 192k OUT.ac3
ffmpeg -i IN.mkv -map 0:1 -c:a ac3 -b:a 192k OUT.ac3
ffmpeg -i IN.mp4 -map 0:1 -c:a ac3 -b:a 192k OUT.ac3

All AC3 with the standard ac3 5 ms delay of silence.

Is there a way to use -pad 0 with ffmpeg?

2.
Audacity don't use the Nero DS decoder to decode AAC, it use ffmpeg.

And in all my test opening IN.aac or IN.mkv or IN.mp4 the decoded aac is perfect, without any cut or delay.

I don't know how you obtain these data. Please explain your workflow.

The workflow is quite straight forward:
eac3to IN.mkv 2: OUT.ac3
Audacity is only used to display the waveforms. Assuming the absence of idiosyncrasy of any kind, it's a simple matter to read off the relative delays between the aac and ac3.

I have uploaded two 10sec samples for your testing pleasure. The first has the remarkable ability to survive Nero's molestation, while the other has a sampling frequency of 48000/24000, which may explain the 42ms truncation instead of 21ms. Perhaps a spectrum analyzer would tell if it is indeed band limited.

Many thanks and best regards.

[Nico8583]

The DTS decoder dcadec (inside ffmpeg and eac3to) is also free soft.

The AC3 encoder inside ffmpeg is better than Aften encoder, both free soft, but maybe there are better certified Dolby Digital commercial encoders.

I say "maybe" because I don't know test about that.
Thank you, I'll look at this.
A last question, why do you use "-center_mixlev 0.707 -surround_mixlev 0.707" with ffmpeg ?

[tebasuna51]

Is there a way to use -pad 0 with ffmpeg?
At least is not ducumented that option:
https://ffmpeg.org/ffmpeg-all.html#ac3-and-ac3_005ffixed

I have uploaded two 10sec samples...

You are right, using your samples I obtain cuts of 0 and 32 ms.
Even with other samples until 54 ms cut.

Then I edited my post http://forum.doom9.org/showthread.php?p=1747412#post1747412 and now must be:

- The directshow Nero 7 decoder, used by eac3to to decode .aac, make unpredictables cuts (from 0 to 54 ms at least) in 2.0 and is broken for 5.1.

But this is still valid:

- eac3to works fine extracting AAC from MKV/TS/M2TS containers, I obtain the same aac than I muxed previously.
Then, to avoid problems, you can use eac3to to extract and Qaac or ffmpeg to decode (or LWLibavAudioSource inside AviSynth).

[tebasuna51]

... why do you use "-center_mixlev 0.707 -surround_mixlev 0.707" with ffmpeg ?

By default ( https://ffmpeg.org/ffmpeg-all.html#ac3-and-ac3_005ffixed ) ffmpeg put -center_mixlev 0.595 -surround_mixlev 0.500:

8.2.1.2 Downmix Levels
----------------------
-center_mixlev level

Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values:

0.707 Apply -3dB gain
0.595 Apply -4.5dB gain (default)
0.500 Apply -6dB gain

-surround_mixlev level

Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. The value is specified as a scale factor. There are 3 valid values:

0.707 Apply -3dB gain
0.500 Apply -6dB gain (default)
0.000 Silence Surround Channel(s)
-center_mixlev 0.707 -surround_mixlev 0.707 is the Aften default.

I recommend use -center_mixlev 0.707 to avoid the low dialog volume when is downmixing to stereo. You can let the default -surround_mixlev or even 0.000 at your preference.

[Nico8583]

Ok thank you, perhaps it could solved dynamic range I can find on DTS track

[mariner]

Thanks for the kind reply, tebasuna51.

1.
At least is not ducumented that option:
https://ffmpeg.org/ffmpeg-all.html#ac3-and-ac3_005ffixed

Thanks for the link. Can ffmpeg add +/- delay like eac3to?

2.

- The directshow Nero 7 decoder, used by eac3to to decode .aac, make unpredictables cuts (from 0 to 54 ms at least) in 2.0 and is broken for 5.1.

Perhaps madshi can be persuaded to consider other candidates for aac decoding?

Many thanks and best regards.

[tebasuna51]

Thanks for the link. Can ffmpeg add +/- delay like eac3to?
You can search at same doc page.
For add +delay: https://ffmpeg.org/ffmpeg-all.html#toc-adelay
To add 500 ms delay to a 6 channel audio add this to command line:

-af adelay=500|500|500|500|500|500

For add -delay: https://ffmpeg.org/ffmpeg-all.html#toc-atrim
To cut fisrt 50 ms to an audio add this to command line:

-af atrim=0.05

Perhaps madshi can be persuaded to consider other candidates for aac decoding?

He don't want add libav aac decoder because license copyright problems.

[jpsdr]

madshi, could we please get an update for dcadec?

Just out of curiosity, can you try and test here (https://mega.nz/#!ZRk1FLxL!tkKYzbipYjTjNgX7unRCCkwkaq3H7lvIARqI-YpDPm4) ?
I've change some compiler options, to make a build, theoricaly, more efficient.
The Intel version is compiled with Intel compiler, and needs a CPU with AVX2 instructions.

[-TiLT-]

Is edit limited to one socond precision? -edit=h:mm:ss,+-delayms
Or is there any way to work with a higher precision? Sth. like -edit:=h:mm:ss:msm,+-delayms?

In several scenaerios, I just want to cut out one or two AC3 frames at a very precise position or insert some silent frames (looping is actually not always an option), but there seem to be no tools out there for such tasks.

[Music Fan]

In several scenaerios, I just want to cut out one or two AC3 frames at a very precise position or insert some silent frames (looping is actually not always an option), but there seem to be no tools out there for such tasks.
Maybe Delaycut.

[LigH]

Or HeadAC3he; but it's rather old, hard to find nowadays.

Oh, look, a signature! :eek:

[tebasuna51]

Is edit limited to one socond precision? -edit=h:mm:ss,+-delayms
Or is there any way to work with a higher precision? Sth. like -edit:=h:mm:ss:msm,+-delayms?

You can use -edit:=h:mm:ss.msm,+-delayms

But remember than eac3to (or DelayCut) only work adding/deleting frames (32 ms for samplerate 48 KHz).
Then the edit point, and delay value, are rounded to near value (precision +- 16 ms)

Is not possible better precision without recode.

[Yoshi]

Due to the circumstances, I encountered yet another movie example where I wonder if the decoding is correctly done by eac3to/ffmpeg or not.

"Everest" comes with a 7.1 TrueHD Atmos track which is, decoded to 7.1 FLAC full of clipping in certain scenes.

For the sake of comparison, I extracted the AC3 core and let that one decode to 5.1 PCM. While eac3to suggests a gain of -6.2dB after encountering clipping and recognizing it as such, the result looks just as bad as the "lossless" one (as I learned, this seems to be a relative term when it comes to multichannel audio in conjunction with TrueHD and DTS-HD MA), only more quiet of course.

Here is twice the left channel of the soundtrack.

http://www.bilder-upload.eu/thumb/35c338-1453249010.jpg (http://www.bilder-upload.eu/show.php?file=35c338-1453249010.jpg)

Now I wonder: is that particular clipping already contained in the mix (and maybe even intended) or is it screwed up during decoding? In other words: how can I be sure which type of clipping I'm dealing with - the immanent one or the artificial one introduced by bugs or difficulty of the decoding itself thanks to the floating point vs. integer dilemma?

[torturesauce]

The foobar2000 developer for the DTS plugin had abandoned support of dcadec and reverted back to the old DTS codec because dca was too slow and buggy at the time. Can somebody please make a fork or something with the latest dcadec so I can run some tests with it?

[tebasuna51]

The foobar2000 developer for the DTS plugin had abandoned support of dcadec and reverted back to the old DTS codec because dca was too slow and buggy at the time. Can somebody please make a fork or something with the latest dcadec so I can run some tests with it?
Buggy for what? Is this the problem?
Problems almost entirely resulting from the switch to an incomplete implementation such as dcadec, which not only did not have any implementation for the common 14 bits data / 2 bits padding per 16 bit word format of most older DTS streams, it also fails to recognize some DTS CDs outright. It's also significantly slower at decoding.

That's don't affect at all to DTS movie trakcs, and the problem can be solved with a data parser to restore the standard 16 bits/word DTS format.
The old BeSplit can do the job without problems.

Maybe this is important for Foobar2000 but not here, and the old DTS decoder foobar plugin don't support DTS-HD at all, at least in my test.

[torturesauce]

Buggy for what? Is this the problem?


That's don't affect at all to DTS movie trakcs, and the problem can be solved with a data parser to restore the standard 16 bits/word DTS format.
The old BeSplit can do the job without problems.

Maybe this is important for Foobar2000 but not here, and the old DTS decoder foobar plugin don't support DTS-HD at all, at least in my test.



Yes, that's it. It had serious issues with DTS CDs. And there is an old foobar plugin for DTS-HD. (http://sourceforge.net/projects/dvdadecoder/files/foo_input_dtshd/) Okay, I guess I'll ask about it on Hydrogenaudio instead of here. Thanks!