Looking for a eac3 7.1 encoder

[tebasuna51]

Can you please complete the table with THD, DTS and PCM up to 7.1?

I add the DTSHD column because some encoders need mono wavs, but I don't know the names for THD encoders and we don't need know at all the internal names and order.

And the most important question: does FFMPEG honor that order when decoding to WAV?

Of course, all actual decoders than I know change the internal codec channel order to the wav order and fill the ChannelMask field of WAVE_FORMAT_EXTENSIBLE header.

And all actual encoders than I know read wav input ChannelMask and create the equivalent internal channel order if is available (if not show a error). If wav input have a simple header (without ChannelMask) assign a default channel map for each number of channels.

For instance for qaac encoder:
In addition to mono/stereo, following channel layouts are supported for AAC and ALAC with slight variations.
When channel layout is unknown, one of the layouts indicated by asterisk is assumed as default.
# chn input layout output layout codec
----------- --- --------------------------------- ----------------------- ------------
3ch* 3 FL FR FC C L R LC, ALAC
4ch 4 FL FR FC BC C L R Cs LC, ALAC
4ch * 4 FL FR BL BR (SL SR) L R Ls Rs LC, HE
5ch * 5 FL FR FC BL BR (SL SR) C L R Ls Rs LC, ALAC
5.1ch * 6 FL FR FC LF BL BR (SL SR) C L R Ls Rs LFE LC, HE, ALAC
6ch 6 FL FR FC BL BR BC (SL SR) C L R Ls Rs Cs LC
6.1ch * 7 FL FR FC LF BL BR BC (SL SR) C L R Ls Rs Cs LFE LC, ALAC
7ch 7 FL FR FC BL BR SL SR C L R Ls Rs Rls Rrs LC
7.1ch front 8 FL FR FC LF BL BR FLC FRC (SL SR) C Lc Rc L R Ls Rs LFE LC, HE, ALAC
7.1ch rear* 8 FL FR FC LF BL BR SL SR C L R Ls Rs Rls Rrs LFE LC, HE
8ch 8 FL FR FC BL BR BC SL SR C L R Ls Rs Rls Rrs Cs LC

(SL SR) is accepted like BL BR to output Ls Rs
Rls Rrs is here MediaInfo Lb Rb (or wav BL BR or EAC3 Lrs Rrs)
Like you see the order and name aren't the same but the encoder do the job atomatically.

[tebasuna51]

I make a test also with TotalCode Studio.

We can configure it like with Audition with some differences:

1) For big files like movie tracks we need rf64 input files (without support for .w64).
And need remap or use the internal remap showed in the image. We can use eac3to:

eac3to 8v341.dtshd 8v341_R.rf64 -0,1,2,3,6,7,4,5

2) The max bitrate is also 1024 Kb/s but there are different channel layouts (see the images).
Without the (L R C LFE Ls Rs Vhl Vhr) than work fine with my audio system 3D 5.2.1.

3) Is fast than Audition.
In my previous old PC (Intel i5-3550 3.30GHz, DDR3 2+2 GB PC3-10700 (667 MHz), SATA ST2000DL003-9VT166 2TB) a source file (Jurasic Park 2h 6m 37s) converted previously to rf64 take 10m 36s to be converted to .ec3.
Audition take 4m to load the .w64 and after 14m to convert it to ec3.

[tormento]

And need remap or use the internal remap showed in the image. We can use eac3to
I prefer to use ffdshow -rf64 auto because it has updated decoders.

Would you please try to use it and tell me if I need channel remapping?

I currently have 2.0 HT here, so I encode to DD+ for archive purpose only and can't directly test.

[tebasuna51]

I prefer to use ffdshow -rf64 auto because it has updated decoders.

Don't work for me:
[NULL @ 000001c14431ee00] Unable to find a suitable output format for 'C:\tmp\8v341.dtshd_.rf64'
C:\tmp\8v341.dtshd_.rf64: Invalid argument

Would you please try to use it and tell me if I need channel remapping?

Of course:
-filter_complex "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c6|c5=c7|c6=c4|c7=c5"

[tormento]

Don't work for me
You need to output a WAV file, as RF64 is just a header.

Such as:

ffmpeg.exe -i whatever.thd -rf64 auto whatever.wav

Try and tell me.

[tebasuna51]

You need to output a WAV file, as RF64 is just a header.

I tried with a small file and just output a wav with RIFF header.

Seems need to be a big file (>4 GB) to create a RF64 header.
No problem, if TotalCode Studio accept .wav with RF64 header, instead the .rf64 extension.

[tormento]

I tried with a small file and just output a wav with RIFF header.
Yes, that is the purpose of the auto parameter.

Unfortunately EAC3TO development is frozen, that is why I suggest to use ffmpeg, at least for the decoding part.

The command line that you gave me was for ffmpeg or EAC3TO? And for TotalCode or Audition?

[tebasuna51]

With ffmpeg:

ffmpeg.exe -i whatever.thd -filter_complex "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c6|c5=c7|c6=c4|c7=c5" -rf64 auto -acodec pcm_s24le whatever.wav

Needed for both, TotalCode or Audition, when the source is a standard 7.1 and the output is remapped to FL FR FC LFE SL SR BL BR in the order to obtain the EAC3 L R C LFE Ls Rs Lrs Rrs

With TotalCode you can do the remap inside the dialog 'Channels' window.

[tebasuna51]

WARNING:

By default the ffmpeg output is downsize to 16 bitdepth, no mather the source bitdepth:

Input #0, dts, from 'D:\Test\MA71.dts':
Stream #0:0: Audio: dts (DTS-HD MA), 48000 Hz, 7.1, s32p (24 bit)
Stream mapping:
Stream #0:0 -> #0:0 (dts (dca) -> pcm_s16le (native))
Output #0, wav, to 'D:\Test\MA71.dts_.wav':
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s16, 6144 kb/s
encoder : Lavc59.18.100 pcm_s16le

eac3to check the input bitdepth, even if is marked wrong, an output the correct bitdepth, you can lose precission with the actualized decoders (without changes for dtsma/thd standard for longtime ago)

With ffmpeg you must know the source bitdepth or add always -acodec pcm_s24le. Now:
Input #0, dts, from 'D:\TestMA71.dts':
Stream #0:0: Audio: dts (DTS-HD MA), 48000 Hz, 7.1, s32p (24 bit)
Stream mapping:
Stream #0:0 -> #0:0 (dts (dca) -> pcm_s24le (native))
Output #0, wav, to 'D:\Test\MA71.dts_.wav':
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s32, 9216 kb/s
encoder : Lavc59.18.100 pcm_s24le

[tebasuna51]

How I do the job (see the image):

1) Load the input file in UsEac3to, and let use the eac3to bitdepth detection

2-4) Select (or write yourself) the desired parameters for ffmpeg

5-6) Click 'A/V Recode' and Replace the decoder by the ffmpeg parameters

7) Select output file 'WAV'
UsEac3to select like encoder "-acodec pcm_s16le" for 16 bit DTSMA or "-acodec pcm_s24le" for 24 bit DTSMA or THD by default.
Many THD are marked as 24 even when are only 16 (detected by eac3to) but if after is recoded to eac3 it is not a problem.

The command line generated is like (without paths to input and output folders):
ffmpeg -i "INPUT" -filter_complex "pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c6|c5=c7|c6=c4|c7=c5" -rf64 auto -acodec pcm_s24le "INPUT_.wav"

[tormento]

Needed for both, TotalCode or Audition, when the source is a standard 7.1
Is it ffmpeg fault or wav format fault in your opinion?

Do we need the same remap for 7.1 DTS, 5.1 THD and 5.1 DTS?

[tebasuna51]

Is a encoder fault than ignore the channelmap stored in the wav header, writed correctly by ffmpeg.

AAC, FLAC and OPUS encoders don't need remap at all.

5.1 don't need remap, 7.1 standard always with these encoders.
The 5.1.2 in Audition don't need remap.

[tormento]

Is a encoder fault than ignore the channelmap stored in the wav header, writed correctly by ffmpeg.
Thanks!

Please explain me how to check if a encoder loses channel configuration and how to understand the proper channel "wiring" to be done.

[tormento]

UsEac3to select like encoder "-acodec pcm_s16le" for 16 bit DTSMA or "-acodec pcm_s24le" for 24 bit DTSMA or THD by default.
What does it choose for already existing EAC3?

Let us say I want to lower bitrate from 896 to 640. Would it use 24 or 16 bits?

Does it need channel remap when 7.1 or is it in the right order as wav from eac3?

[tebasuna51]

What does it choose for already existing EAC3?

Let us say I want to lower bitrate from 896 to 640. Would it use 24 or 16 bits?
Only lossless encodes (TrueHD, DTS-MA, FLAC,...) have a bitdepth

Lossy encodes don't have bitdepth. Decoders work internally in float format, eac3to can output until 64 bits float, but it is enough 24 bits int (human ear can difference only to 20 bits).

Does it need channel remap when 7.1 or is it in the right order as wav from eac3?

The correct wav order is always the same, no mather the source codec, then always must be remapped for that encoders.
Thanks!

Please explain me how to check if a encoder loses channel configuration and how to understand the proper channel "wiring" to be done.

Download the channel test in https://forum.doom9.org/showthread.php?p=1963170#post1963170

If you don't have a 7.1 system to listen you always can "see" the order, the first image is the correct the second is a wrong order:

[tormento]

Download the channel test
Thanks!

Do you know some audio editor that can open THD, DTSHD and so?

What is that one?

[tebasuna51]

Audo editors only work with uncompressed audio.

Decode anything to wav/w64 before edit.

I work with Goldwave from longtime ago, but you can use Audacity also.

[tormento]

Many THD are marked as 24 even when are only 16 (detected by eac3to) but if after is recoded to eac3 it is not a problem.
Is there a way to find the correct bitdepth?

[tebasuna51]

Let eac3to decode it.
It analyze all decoded samples to see how many bits have info.

[tormento]

Let eac3to decode it.
I have found a (I think) strange 6.1 (7?) audio DTS from Ghost in the shell 2.0 BD.

mediainfo:
Audio #1
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : DTS ES XLL
Format/Info : Digital Theater Systems
Commercial name : DTS-HD Master Audio
Muxing mode : Stream extension
Codec ID : 134
Duration : 1 h 23 min
Bit rate mode : Variable / Constant / Constant
Bit rate : Unknown / 1 509 kb/s / 1 509 kb/s
Channel(s) : 7 channels / 7 channels / 6 channels
Channel layout : C L R Ls Rs LFE Cb / C L R Ls Rs Cb LFE / C L R Ls Rs LFE
Sampling rate : 48.0 kHz
Frame rate : 93.750 FPS (512 SPF)
Bit depth : 24 bits
Compression mode : Lossless / Lossy / Lossy
Language : Italian
Source : 00009.m2ts

Audio #2
ID : 4353 (0x1101)
Menu ID : 1 (0x1)
Format : DTS ES XXCH XLL
Format/Info : Digital Theater Systems
Commercial name : DTS-HD Master Audio
Muxing mode : Stream extension
Codec ID : 134
Duration : 1 h 23 min
Bit rate mode : Variable / Constant / Constant
Bit rate : Unknown / 1 509 kb/s / 1 509 kb/s
Channel(s) : 7 channels / 7 channels / 6 channels
Channel layout : C L R Ls Rs LFE Cb / C L R Ls Rs Cb LFE / C L R Ls Rs LFE
Sampling rate : 48.0 kHz
Frame rate : 93.750 FPS (512 SPF)
Bit depth : 16 bits
Compression mode : Lossless / Lossy / Lossy
Language : Japanese
Source : 00009.m2ts

eac3to:

DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 1509kbps, 48kHz)

DTS Master Audio, 6.1 channels, 16 bits, 48kHz
(core: DTS-ES, 6.1 channels, 1509kbps, 48kHz)
Can you suggest me the correct decoding channel parameters for them, in ffmpeg format?

[SeeMoreDigital]

I have found a (I think) rare 6.1 audio DTS from Ghost in the shell 2.0 BD.

mediainfo:
General
Complete name : F:\In\1_26 Ghost in the shell 2.0 (Dynit)\[jpn]6116.dts
Format : DTS ES XXCH XLL
Format/Info : Digital Theater Systems
Commercial name : DTS-HD Master Audio
File size : 1.26 GiB
Overall bit rate mode : Variable / Constant / Constant

Audio
Format : DTS ES XXCH XLL
Format/Info : Digital Theater Systems
Commercial name : DTS-HD Master Audio
Bit rate mode : Variable / Constant / Constant
Bit rate : Unknown / 1 509 kb/s / 1 509 kb/s
Channel(s) : 7 channels / 7 channels / 6 channels
Channel layout : C L R Ls Rs LFE Cb / C L R Ls Rs Cb LFE / C L R Ls Rs LFE
Sampling rate : 48.0 kHz
Frame rate : 93.750 FPS (512 SPF)
Bit depth : 16 bits
Compression mode : Lossless / Lossy / Lossy

eac3to:

DTS Master Audio, 6.1 channels, 16 bits, 48kHz
(core: DTS-ES, 6.1 channels, 1509kbps, 48kHz)
Can you suggest me the correct decoding channel parameters for it, in ffmpeg format?
Have you extracted the lossy DTS-ES core from the lossless DTS-HD MA audio stream?

[tormento]

Have you extracted the lossy DTS-ES core from the lossless DTS-HD MA audio stream?
Nope, just demuxed the 2 streams from m2ts.

I think eac3to went nuts. :o

I forgot to say: I need the wave remapping (if any) to have the correct 6.1 (or whatever) DD+ TotalCode Studio (or Audition) encoding.

ffprobe gives me:
Stream #0:0: Audio: dts (DTS-HD MA), 48000 Hz, 5.1(side), s32p (24 bit)

Stream #0:0: Audio: dts (DTS-HD MA), 48000 Hz, 6.1, s16p

[tebasuna51]

You don't need nothing to remap for the first one 5.1

The second one (16 bits) is decoded to 6.1 (FL FR FC LFE BC SL SR) and if you want to encode to EAC3 6.1 (L,R,C,LFE,Ls,Rs,Cs) you must remap the BC in wav to the last position:

ffmpeg -i "[jpn]6116.dts" -filter_complex "pan=6.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c5|c5=c6|c6=c4" -acodec pcm_s16le "6.1_Channels_R.w64"

for Audition or, for TotalCode Studio with
-rf64 auto -acodec pcm_s16le "6.1_Channels_R.wav"

[tormento]

for Audition or, for TotalCode Studio with
-rf64 auto -acodec pcm_s16le "6.1_Channels_R.wav"
Thanks a lot.

I was literally waiting for your help :)

[tebasuna51]

DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 1509kbps, 48kHz)


Seems the first one [ita] can be (DTS-ES) a matrixed 6.1, there are only 5.1 physical channels but maybe you can enable the EX flag when encode the 5.1 (without remap).

In Audition: BitStream Information -> Dolby Digital Surround EX mode -> Enabled
Maybe TotalCode Studio can set that flag also, I don't remember.

[pcroland]

I created a wrapper for DEE's DD, DDP and TrueHD encoder: Doom9 (https://forum.doom9.org/showthread.php?p=1969949), GitHub (https://github.com/pcroland/deew)

[tormento]

I created a wrapper for DEE's DD, DDP and TrueHD encoder
That is awesome.

I have tried and it works flawlessly.

[tormento]

@tebasuna51

I was thinking about using Dolby Reference Player to properly decode audio. Did you ever used it? I think it should be launched from CLI for the decoding part.

[tebasuna51]

I will test it but I don't have much time now.

[Enverex]

My god, I can finally post.

Anyway, I got the Plex DDP encoder working standalone and creating 7.1 content. Everything works great and correct, except for ONE issue. It appears to be adding forced processing to the audio in some way - it sounds like dialogue enhancement or something. Every time there's audio in the center channel (typically dialogue) it abruptly ducks the volume of the surround speakers and unducks it the moment the center channel goes quiet again.

No dynamic compression, enhancement, etc is enabled on the listening end, so this effect seems like it's getting hard baked into the audio and is ruining what would otherwise be perfect 7.1 encodes.

Anyone know why this may be happening and how to stop EAE from doing it?

[tebasuna51]

... it sounds like dialogue enhancement or something. Every time there's audio in the center channel (typically dialogue) it abruptly ducks the volume of the surround speakers and unducks it the moment the center channel goes quiet again.
It is well know than Plex encoder add Dynamic Range Compression to the eac3 tracks, and I don't know how avoid that.

No dynamic compression, enhancement, etc is enabled on the listening end,
By default the decoders must obey the DRC info and the players must disable explicitly the DRC to avoid that effect.
What is the player used?

[Balling]

It is well know than Plex encoder add Dynamic Range Compression to the eac3 tracks, and I don't know how avoid that.


By default the decoders must obey the DRC info and the players must disable explicitly the DRC to avoid that effect.
What is the player used?

The way it is done in ffmpeg decoder itself is -drc_scale 0. You do not need to avoid that in encoding. See https://patchwork.ffmpeg.org/project/ffmpeg/patch/20200201193443.22419-1-rcombs@rcombs.me/


Ffmpeg's encoder has never supported generating it.

[pcroland]

It is well know than Plex encoder add Dynamic Range Compression to the eac3 tracks, and I don't know how avoid that.


By default the decoders must obey the DRC info and the players must disable explicitly the DRC to avoid that effect.
What is the player used?

With DEE you can only choose from these DRC settings:
['film_light', 'film_standard', 'music_light', 'music_standard', 'speech']
You can't disable it in any way.

When it comes to dialnorm, you can set it between -31 and -1dB, and 0 will automatically calculate it (and you can't skip measrement even if you set it manually). You can also set a metering_mode and a speech_threshold for it, but they produce bit identical files no matter what you set these.

[Balling]

With DEE you can only choose from these DRC settings:
['film_light', 'film_standard', 'music_light', 'music_standard', 'speech']
You can't disable it in any way.

When it comes to dialnorm, you can set it between -31 and -1dB, and 0 will automatically calculate it (and you can't skip measrement even if you set it manually). You can also set a metering_mode and a speech_threshold for it, but they produce bit identical files no matter what you set these.

'You can't disable it in any way."

The whole point of DRC (not really dialnorm, that is separate) is that you can remove the compression if you do not care about your neighbours.

[pcroland]

I know you can disable it in other encoders but not with DEE for some reason.

[tebasuna51]

I know you can disable it in other encoders but not with DEE for some reason.
If DEE can't use the None preset at least use always the Music Light preset.
DRC six presets available to choose from, including None.

Music Light Music Standard Film Light Film Standard Speech
dB range (ratio) dB range (ratio) dB range (ratio) dB range (ratio) dB range (ratio)
---------- ------- ---------- ------- ---------- ------- ---------- ------- ---------- -------
Max Boost (fix): < -65 (12 dB) < -55 (12 dB) < -53 ( 6 dB) < -43 ( 6 dB) < -50 (15 dB)
Boost Range: -65 to -41 (2:1) -55 to -31 (2:1) -53 to -41 (2:1) -43 to -31 (2:1) -50 to -31 (2:1)
Null Band Width: -41 to -21 (1:1) -31 to -26 (1:1) -41 to -21 (1:1) -31 to -26 (1:1) -31 to -26 (1:1)
Early Cut Range: nothing -26 to -16 (2:1) -21 to -11 (2:1) -26 to -16 (2:1) -26 to -16 (2:1)
Cut Range: -21 to +9 (2:1) -16 to +4 (20:1) -11 to +4 (20:1) -16 to +4 (20:1) -16 to +4 (20:1)

Not problem with Max Boost below -65 dB, the big Null Band.
And without the ratio 20:1, the culprit of abrupt volume changes.

[Balling]

I know you can disable it in other encoders but not with DEE for some reason.

I was talking about decoder. You can disable it in a decoder. DRC is required to get good quality in AN ENCODER. FFmpeg not having it may be one of key reasons it perfoms bad for eac3.

[Balling]

If you encode flac to eac3 to get decoder close to flac you need to turn off DRC in the decoder. See this audacity patch, that uses ffmpeg and requires drc_scale 0. Last comment is mine: https://github.com/audacity/audacity/issues/631

[Balling]

Only lossless encodes (TrueHD, DTS-MA, FLAC,...) have a bitdepth

Lossy encodes don't have bitdepth. Decoders work internally in float format, eac3to can output until 64 bits float, but it is enough 24 bits int (human ear can difference only to 20 bits).





TrueHD is always 24 bit, it does not have a bitdepth by your definition. Flac can be even 8 bit.

[SeeMoreDigital]

TrueHD is always 24 bit, it does not have a bitdepth by your definition. A fair amount has been padded from 16 to 24-bit!

Flac can be even 8 bit.Maybe, but only if the source was 8-bit.... But who would do that? Certainly no commercial provider!

[tebasuna51]

... DRC is required to get good quality in AN ENCODER. FFmpeg not having it may be one of key reasons it perfoms bad for eac3.
The metadata DRC can help for old and slow decoders, but with modern and fast decoders the metadata in DD encodes aren't needed at all.
There are filters like 'night mode' than can do the job on the fly over any codec.
For that ffmpeg not need at all create those unnecesary DRC metadata.

[Balling]

A fair amount has been padded from 16 to 24-bit!

Maybe, but only if the source was 8-bit.... But who would do that? Certainly no commercial provider!

Yes, but there is no way to tell that before decoding, okay? The metadata is not there, which means you should decode always to 24 bits. The variant of TrueHD called MLP does have such metadata.

[pcroland]

If DEE can't use the None preset at least use always the Music Light preset.
Changed the default DRC to music_light in deew, thanks for the info.
deew 2.6.0:
- changed default DRC to music_light, which is closer to the missing
none preset: https://forum.doom9.org/showpost.php?p=1972689&postcount=136
- fixed DRC option not being applied for stereo TrueHD encodes
- colorized argparse's help a little bit
- renamed -t/--threads to -in/--instances to be more accurate,
since one DEE instance can use 2 threads. see -h for more info
I'm thinking about implementing the none preset with a binary edit. Do you know which bits should be replaced with what? If i recall correctly you have to do that for every frame, right?

[tebasuna51]

...
I'm thinking about implementing the none preset with a binary edit. Do you know which bits should be replaced with what? If i recall correctly you have to do that for every frame, right?
Read the E/AC3 syntax here (https://web.archive.org/web/20110626050743/http://www.atsc.org/cms/standards/a_52-2010.pdf)

Each frame have 6 Audio Blocks (point 5.2 in pdf) and for each Audio Block there are the DRC values (7.7): "The bit code of ‘0000 0000’ indicates 0 dB (unity) gain"

You must replace by 0's in each Audio Block in each audio frame, the problem is locate these bits. You must know the size of bsi (5.3) and the audblk syntax (5.4.3)

[pcroland]

So it's not that easy to implement. Do you know a working tool for it by any chance?

[tebasuna51]

No, sorry.

[BuccoBruce]

Yes, the sample I created in that post was using the "audiotool" I wrote here (https://github.com/ymgenesis/AudioTool). The command would be the same as I quoted in the Plex forum post. I don't believe an internet connection is needed. On Mac, it requires the structure from here (https://github.com/ymgenesis/AudioTool#dependencies-and-folder-structure). Variables for the Codecs and Encoder folder need to be set properly (folder with dylibs, probably dlls in Windows, and the Plex Transcoder executable). It needs EasyAudioEncoder executable and the license file. This license file is the permission file to be able to use EasyAudioEncoder executable, so I don't think an internet connection is needed. Keep in mind all these resources (including EasyAudioEncoder and its license file) are insalled when installing Plex. They get updated every once and a while when Plex updates. EasyAudioEncoder must be running for the encode to work, as you can see I execute EasyAudioEncoder first in the background here (https://github.com/ymgenesis/AudioTool/blob/904d3f56abd13e01857a686deb611a0ef8103e8f/audiotool.sh#L329) before executing the Plex Transcoder executable here (https://github.com/ymgenesis/AudioTool/blob/904d3f56abd13e01857a686deb611a0ef8103e8f/audiotool.sh#L331) a couple lines later. Of course, this is Unix.

Sorry I can't be more help with Windows. The process isn't exactly clean, as it requires the proper environment variables, and needs a running EasyAudioEncoder instance before executing the Plex Transcoder (ffmpeg) command.

And as I mentioned before, I paid for Plex Pass, which may have allowed me EasyAudioEncoder for 7.1 transcoding when playing on Plex (existence of EasyAudioEncoder and license file). BUT, I don't know that 100%, I actually suspect it is included with the free version. So, if someone wants to try, they can sign up for a free Plex account at plex.tv and download my sample from here (https://www.mediafire.com/file/sicv8mizm2tsnov/sample.mkv/file) (same sample as on the Plex forums), add it to your server once setup, and see if it'll transcode the first audio track (TrueHD) to EAC3 7.1 when playing back on your 7.1-capable TV/box & AV setup. If it does, EasyAudioEncoder exists and can be used in a manner like my "audiotool".

If you can explain which environment variables EAE/Plex Transcoder actually require, and what conditions they need in order to avoid halting them midway through, I can re-write this properly to work under Windows.

I don't understand enough about how .sh scripts work to properly re-write it for Windows (bc I don't know what it's doing most of the time) but I was able to somewhat successfully run this under msys2 in Windows with some re-writes. The access to out.txt is still throwing "safe_fopen: CreateFileW failed: 2" errors, but it does seem to work. With the "| tee" to out.txt still included in the thd/dts functions, it was halting the encode after a few seconds (a few minutes of output). Without them, it finishes the encode, although the time-left function is now buggered.

frame= 0 fps=0.0 q=-1.0 size= 285755kB time=00:05:52.67 bitrate=6637.6kbits/
safe_fopen: CreateFileW failed: 2
safe_fopen: CreateFileW failed: 2
safe_fopen: CreateFileW failed: 2
frame= 0 fps=0.0 q=-1.0 Lsize= 296962kB time=00:05:53.69 bitrate=6878.0kbits/s speed=4.94x

Terminating EasyAudioEncoder...
Terminated

https://pastebin.com/LDPXXmm2

Required ??? folder structure:
\\AudioTool\audiotool.sh
\\AudioTool\audiotool.win.sh
\\AudioTool\Codecs\367b3d4-3654-windows-x86\aac_decoder.dll
snip
\\AudioTool\Codecs\367b3d4-3654-windows-x86\wmv3_decoder.dll
\\AudioTool\Codecs\ffmpeg.exe
\\AudioTool\Codecs\ffprobe.exe
\\AudioTool\Encoder\eae-license.txt
\\AudioTool\Encoder\EasyAudioEncoder.exe
\\AudioTool\Encoder\ffmpeg.exe
\\AudioTool\Encoder\ffprobe.exe
\\AudioTool\Encoder\Windows\avcodec-58.dll
\\AudioTool\Encoder\Windows\avfilter-7.dll
\\AudioTool\Encoder\Windows\avformat-58.dll
\\AudioTool\Encoder\Windows\avutil-56.dll
\\AudioTool\Encoder\Windows\cyggcc_s-1.dll
\\AudioTool\Encoder\Windows\cygwin1.dll
\\AudioTool\Encoder\Windows\fmt.dll
\\AudioTool\Encoder\Windows\Plex Transcoder.exe
\\AudioTool\Encoder\Windows\swresample-3.dll
\\AudioTool\Encoder\Windows\swscale-5.dll
\\AudioTool\ffmpeg.exe
\\AudioTool\ffprobe.exe
\\AudioTool\README.md


ffmpeg/ffprobe are in every folder because with ffprobe only in \AudioTool, msys couldn't find it, even though .\ffprobe.exe outside of the script was working. They don't come with ffmpeg, I used statically compiled ones separately.

[BuccoBruce]

PSA to anyone using Plex Transcoder/EasyAudioEncoder - Plex's ffmpeg implementation / DCA decoder are not decoding DTS-HD MA 7.1. If I were a betting man, I would bet they're decoding the core DTS 5.1 audio. I can double check to confirm with a "known" decode of the core DTS track using eac3to and then comparing the PCM data, but given that EAE seems to be applying some unwanted DRC, I doubt I'll bother. My university copy of Audition 2015.2 still seems to work even though it complains about a serial number.

Q:\AudioTool\Encoder\Windows>"Plex Transcoder.exe" -c:v:0 h264 -c:a:0 dca -noaccurate_seek -analyzeduration 20000000 -probesize 20000000 -i "Q:\AudioTool\Hawaiian.Vacation.2011.mkv" -c:a pcm_s24le -vn -sn -avoid_negative_ts disabled -hide_banner -start_at_zero -stats -v debug "Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv"
Splitting the commandline.
Reading option '-c:v:0' ... matched as option 'c' (codec name) with argument 'h264'.
Reading option '-c:a:0' ... matched as option 'c' (codec name) with argument 'dca'.
Reading option '-noaccurate_seek' ...Rescanning for external libs: 'Q:/AudioTool/Codecs/367b3d4-3654-windows-x86/'
Loading external lib Q
Error loading external lib: The specified module could not be found.
Loading external lib /AudioTool/Codecs/367b3d4-3654-windows-x86/aac_decoder.dll
/snip/
Loading external lib /AudioTool/Codecs/367b3d4-3654-windows-x86/wmv3_decoder.dll
matched as option 'accurate_seek' (enable/disable accurate seeking with -ss) with argument 0.
Reading option '-analyzeduration' ... matched as AVOption 'analyzeduration' with argument '20000000'.
Reading option '-probesize' ... matched as AVOption 'probesize' with argument '20000000'.
Reading option '-i' ... matched as input url with argument 'Q:\AudioTool\Hawaiian.Vacation.2011.mkv'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_s24le'.
Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'.
Reading option '-sn' ... matched as option 'sn' (disable subtitle) with argument '1'.
Reading option '-avoid_negative_ts' ... matched as AVOption 'avoid_negative_ts' with argument 'disabled'.
Reading option '-hide_banner' ... matched as option 'hide_banner' (do not show program banner) with argument '1'.
Reading option '-start_at_zero' ... matched as option 'start_at_zero' (shift input timestamps to start at 0 when using copyts) with argument '1'.
Reading option '-stats' ... matched as option 'stats' (print progress report during encoding) with argument '1'.
Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'.
Reading option 'Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option hide_banner (do not show program banner) with argument 1.
Applying option start_at_zero (shift input timestamps to start at 0 when using copyts) with argument 1.
Applying option stats (print progress report during encoding) with argument 1.
Applying option v (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url Q:\AudioTool\Hawaiian.Vacation.2011.mkv.
Applying option c:v:0 (codec name) with argument h264.
Applying option c:a:0 (codec name) with argument dca.
Applying option noaccurate_seek (enable/disable accurate seeking with -ss) with argument 0.
Successfully parsed a group of options.
Opening an input file: Q:\AudioTool\Hawaiian.Vacation.2011.mkv.
[NULL @ 0081ec00] Opening 'Q:\AudioTool\Hawaiian.Vacation.2011.mkv' for reading
[file @ 00812cc0] Setting default whitelist 'file,crypto'
[matroska,webm @ 0081ec00] Format matroska,webm probed with size=2048 and score=100
st:0 removing common factor 1000000 from timebase
st:1 removing common factor 1000000 from timebase
st:2 removing common factor 1000000 from timebase
st:3 removing common factor 1000000 from timebase
st:4 removing common factor 1000000 from timebase
st:5 removing common factor 1000000 from timebase
[matroska,webm @ 0081ec00] Before avformat_find_stream_info() pos: 6676 bytes read:35388 seeks:2 nb_streams:6
[h264 @ 00827100] nal_unit_type: 7(SPS), nal_ref_idc: 3
snip
[h264 @ 00827100] ct_type:0 pic_struct:0
Last message repeated 1 times
[matroska,webm @ 0081ec00] All info found
[matroska,webm @ 0081ec00] After avformat_find_stream_info() pos: 1476424 bytes read:1505136 seeks:2 frames:163
Input #0, matroska,webm, from 'Q:\AudioTool\Hawaiian.Vacation.2011.mkv':
Metadata:
title : Hawaiian Vacation (2011)
encoder : libebml v1.3.0 + libmatroska v1.4.1
creation_time : 2014-08-18T18:18:23.000000Z
Duration: 00:05:53.73, start: 0.000000, bitrate: 30402 kb/s
Chapter #0:0: start 0.000000, end 161.453000
Metadata:
title : 00:00:00.000
Chapter #0:1: start 161.453000, end 248.832000
Metadata:
title : 00:02:41.453
Chapter #0:2: start 248.832000, end 353.728000
Metadata:
title : 00:04:08.832
Stream #0:0, 25, 1/1000: Video: h264 (High), 1 reference frame, yuv420p(tv, bt709, progressive, left), 1920x1080 (1920x1088) blah blah
Metadata:
title : MPEG-4 AVC Video / 24884 kbps / 1080p / 23,976 fps / 16:9 / High Profile 4.1
BPS : 24664310
BPS-eng : 24664310
DURATION : 00:05:53.687000000
snip
Stream #0:1(eng), 102, 1/1000: Audio: dts (DTS-HD MA), 48000 Hz, 7.1, s32p (24 bit), 1536 kb/s (default)
Metadata:
title : English / DTS-HD Master Audio / 7.1 / 48 kHz / 5370 kbps / 24-bit (DTS Core: 5.1 / 48 kHz / 1509 kbps / 24-bit)
BPS : 5372315
BPS-eng : 5372315
DURATION : 00:05:53.696000000
snip
Stream #0:2(eng), 33, 1/1000: Audio: ac3, 48000 Hz, stereo, s16p, 320 kb/s
Metadata:
title : Commentary / Dolby Digital Audio / 2.0 / 48 kHz / 320 kbps
snip
Stream #0:3(eng), 2, 1/1000: Subtitle: hdmv_pgs_subtitle, 1920x1080
blah blah blah more subtitles
Successfully opened the file.
Parsing a group of options: output url Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv.
Applying option c:a (codec name) with argument pcm_s24le.
Applying option vn (disable video) with argument 1.
Applying option sn (disable subtitle) with argument 1.
Successfully parsed a group of options.
Opening an output file: Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv.
File 'Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv' already exists. Overwrite ? [y/N] y
[file @ 038e06c0] Setting default whitelist 'file,crypto'
Successfully opened the file.
Stream mapping:
Stream #0:1 -> #0:0 (dts (dca) -> pcm_s24le (native))
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
Last message repeated 1 times
detected 24 logical cores
[graph_0_in_0_1 @ 0083e340] Setting 'time_base' to value '1/48000'
[graph_0_in_0_1 @ 0083e340] Setting 'sample_rate' to value '48000'
[graph_0_in_0_1 @ 0083e340] Setting 'sample_fmt' to value 'fltp'
[graph_0_in_0_1 @ 0083e340] Setting 'channel_layout' to value '0x60f'
[graph_0_in_0_1 @ 0083e340] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x60f
[format_out_0_0 @ 0083e600] Setting 'sample_fmts' to value 's32'
[auto_resampler_0 @ 0083d980] Setting 'output_sample_bits' to value '24'
[format_out_0_0 @ 0083e600] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 0397ffc0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto_resampler_0 @ 0083d980] [SWR @ 02d7a340] Using fltp internally between filters
[auto_resampler_0 @ 0083d980] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:6 chl:5.1(side) fmt:s32 r:48000Hz
Output #0, matroska, to 'Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv':
Metadata:
blah blah
Stream #0:0(eng), 0, 1/1000: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1(side), s32, 6912 kb/s (default)
Metadata:
blah

I noticed when I ran the same AudioEncoder.sh script with -v debug:

"Plex Transcoder.exe" -c:v:0 h264 -c:a:0 dca -noaccurate_seek -analyzeduration 20000000 -probesize 20000000 -i "Q:\AudioTool\Hawaiian.Vacation.2011.mkv" -filter_complex "[0:a:0] aresample=async=1:ocl='7.1':rematrix_maxval=60.000000dB:osr=48000[0]" -map [0] -metadata:s:a title="EAC3 7.1 from DTS" -metadata:s:a language=eng -c:a eac3_eae -b:a 1280k -vn -sn -avoid_negative_ts disabled -hide_banner -start_at_zero -stats -v debug "Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv"
[file @ 03921640] Setting default whitelist 'file,crypto'
Successfully opened the file.
Stream mapping:
Stream #0:1 (dca) -> aresample
aresample -> Stream #0:0 (eac3_eae)
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
Last message repeated 1 times
detected 24 logical cores
[Parsed_aresample_0 @ 02781880] Setting 'async' to value '1'
[Parsed_aresample_0 @ 02781880] Setting 'ocl' to value '7.1'
[Parsed_aresample_0 @ 02781880] Setting 'rematrix_maxval' to value '60.000000dB'
[Parsed_aresample_0 @ 02781880] Setting 'osr' to value '48000'
[graph_0_in_0_1 @ 027819c0] Setting 'time_base' to value '1/48000'
[graph_0_in_0_1 @ 027819c0] Setting 'sample_rate' to value '48000'
[graph_0_in_0_1 @ 027819c0] Setting 'sample_fmt' to value 'fltp'
[graph_0_in_0_1 @ 027819c0] Setting 'channel_layout' to value '0x60f'
[graph_0_in_0_1 @ 027819c0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x60f
[format_out_0_0 @ 02781700] Setting 'sample_fmts' to value 'flt|s16'
[format_out_0_0 @ 02781700] Setting 'sample_rates' to value '48000'
[format_out_0_0 @ 02781700] Setting 'channel_layouts' to value '0x60f|0x63f'
[AVFilterGraph @ 039edfc0] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
[Parsed_aresample_0 @ 02781880] picking flt out of 2 ref:fltp
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] Using fltp internally between filters
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] Matrix coefficients:
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] FL: FL:1.000000 FR:0.000000 FC:0.000000 LFE:0.000000 SL:0.000000 SR:0.000000
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] FR: FL:0.000000 FR:1.000000 FC:0.000000 LFE:0.000000 SL:0.000000 SR:0.000000
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] FC: FL:0.000000 FR:0.000000 FC:1.000000 LFE:0.000000 SL:0.000000 SR:0.000000
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] LFE: FL:0.000000 FR:0.000000 FC:0.000000 LFE:1.000000 SL:0.000000 SR:0.000000
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] BL: FL:0.000000 FR:0.000000 FC:0.000000 LFE:0.000000 SL:0.000000 SR:0.000000
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] BR: FL:0.000000 FR:0.000000 FC:0.000000 LFE:0.000000 SL:0.000000 SR:0.000000
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] SL: FL:0.000000 FR:0.000000 FC:0.000000 LFE:0.000000 SL:1.000000 SR:0.000000
[Parsed_aresample_0 @ 02781880] [SWR @ 02de9fc0] SR: FL:0.000000 FR:0.000000 FC:0.000000 LFE:0.000000 SL:0.000000 SR:1.000000
[Parsed_aresample_0 @ 02781880] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:8 chl:7.1 fmt:flt r:48000Hz
Output #0, matroska, to 'Q:\AudioTool\Hawaiian.Vacation.2011.out.mkv':

If I add icl='7.1' and try to decode to PCM, it still only detects 5.1
Stream mapping:
Stream #0:1 (dca) -> aresample
aresample -> Stream #0:0 (pcm_s24le)
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
Last message repeated 1 times
detected 24 logical cores
[Parsed_aresample_0 @ 026f9900] Setting 'async' to value '1'
[Parsed_aresample_0 @ 026f9900] Setting 'icl' to value '7.1'
[Parsed_aresample_0 @ 026f9900] Setting 'rematrix_maxval' to value '60.000000dB'
[Parsed_aresample_0 @ 026f9900] Setting 'osr' to value '48000'
[graph_0_in_0_1 @ 026fa240] Setting 'time_base' to value '1/48000'
[graph_0_in_0_1 @ 026fa240] Setting 'sample_rate' to value '48000'
[graph_0_in_0_1 @ 026fa240] Setting 'sample_fmt' to value 'fltp'
[graph_0_in_0_1 @ 026fa240] Setting 'channel_layout' to value '0x60f'
[graph_0_in_0_1 @ 026fa240] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x60f
[format_out_0_0 @ 026fa100] Setting 'sample_fmts' to value 's32'
[AVFilterGraph @ 03950b40] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
[Parsed_aresample_0 @ 026f9900] [SWR @ 02d4a300] Using fltp internally between filters
[Parsed_aresample_0 @ 026f9900] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:6 chl:5.1(side) fmt:s32 r:48000Hz

And the ffmpeg binary Plex built from is ancient:
Q:\AudioTool\Encoder\Windows>"Plex Transcoder.exe" -codecs
ffmpeg version 1.6-367b3d4c85-1 Copyright (c) 2000-2019 the FFmpeg developers
D.A.LS dts DCA (DTS Coherent Acoustics) (decoders: dca )
libavutil 56. 26.100 / 56. 26.100
libavcodec 58. 52.100 / 58. 52.100
libavformat 58. 27.104 / 58. 27.104
libavfilter 7. 49.100 / 7. 49.100
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100

It's loading its DTS decoder externally from "dca_decoder.dll" in the codecs folder. It's probably worth mentioning that I am deliberately using an old as hell Plex version, because I got sick of them breaking AMD hardware transcoding...

Yes, the sample I created in that post was using the "audiotool" I wrote here (https://github.com/ymgenesis/AudioTool).

I'm not sure if your aresample filtergraph is necessary? I might try it with a 7.1 pcm_s24le source later, but I may not bother. If your installation of Plex is up to date, double check that the transcoder is actually decoding DTS-HD MA in 7.1.

I almost have this running under Windows natively, but it halts on:
[eac3_eae @ 02764cc0] got broken frame from EAE
Audio encoding failed

These were the environment variables I had to change. If you're on Windows and you don't want to risk this stuff persisting afterwards, just run "SETLOCAL" first in cmd.exe

SETLOCAL
SET EAE_ROOT=Q:\AudioTool\Encoder
SET FFMPEG_EXTERNAL_LIBS=Q:/AudioTool/Codecs/367b3d4-3654-windows-x86/
SET X_PLEX_TOKEN=xxxxxxxxxxxxxxxxxxxx

EAE itself is in another cmd window spitting out endless errors just like it did in msys2, but the encode actually worked from there.
Q:\AudioTool\Encoder>EasyAudioEncoder.exe
Dolby, Dolby Digital, Dolby Digital Plus, Dolby TrueHD and the double D symbol are trademarks of Dolby Laboratories.
safe_fopen: CreateFileW failed: 2
safe_fopen: CreateFileW failed: 2
safe_fopen: CreateFileW failed: 2 ad infinitum

[BuccoBruce]

It's loading its DTS decoder externally from "dca_decoder.dll" in the codecs folder. It's probably worth mentioning that I am deliberately using an old as hell Plex version, because I got sick of them breaking AMD hardware transcoding...

I'm not sure if your aresample filtergraph is necessary? I might try it with a 7.1 pcm_s24le source later, but I may not bother. If your installation of Plex is up to date, double check that the transcoder is actually decoding DTS-HD MA in 7.1.


For what it's worth, I did not find any meaningful change log entries for DTS decoding, 7.1 channels, etc. between my currently installed version of PMS and their latest release https://forums.plex.tv/t/plex-media-server/30447/ DTS isn't mentioned after July of 2019. https://files.catbox.moe/v715bz.png emphasis mine :scared:

Perhaps they should focus on basic A/V functionality, like AV1 support and proper DTS-HD MA decoding, before they add more u$ele$$ feature$ like TIDAL, vidja, and "free" shovelware from the media conglomerates ;) :D wish I'd never paid them a dime...

[BuccoBruce]

Yep. It's only decoding the core. I found a creative way to "force" a test,


Get vuvuzela sounds
Get a 5.1 source
Mix the vuvuzelas in as Lsr/Rsr
Invert the vuvuzelas, add them to the pre-existing audio for Lss/Rss
Confirm your 5.1 downmix settings have both Lss/Lsr at 0 dB for the Ls downmix and Rss/Rsr 0dB for Rs downmix. The opposite rear surround should be set to -INF. Do not invert anything, you've already done so in your cursed source audio. This dictates how the DTS Core stream is created as well!
?
Profit


https://files.catbox.moe/meaped.png

https://files.catbox.moe/8s2el9.png

It goes without saying that there are vuvuzelas on this, so turn your speakers down:

DIDNEYWORL.mkv (https://mega.nz/file/DjJkGbBA#4i4wjY_rf4WuiK1NuAnq-1_6mSAexXvpwqVejesDxxM)


DTS-HD MA 7.1 - "Cursed Castle" feat. Vuvuzelas
Extracted DTS Core - Side/Rear Vuvuzelas Canceled Out!
Plex/EAE E-AC-3 - No Vuvuzelas!
eac3to.exe DIDNEYVOOVINVF.dtshd DIDNEYVOOVINVF.dtshd_.flac
eac3to -0,1,2,3,6,7,4,5 FLAC -> E-AC-3 (Adobe Audition 2015.2) = BZZZZZZZZZZZZZZZZZZZZZZZZZZZZZZZZZZZ


Audio #1

ID : 2
Format : DTS XLL
Format/Info : Digital Theater Systems
Commercial name : DTS-HD Master Audio
Codec ID : A_DTS
Duration : 15s 595ms
Bit rate mode : VBR
Bit rate : 4433 Kbps
Channel(s) : 8 channel
Channel layout : C L R LFE Lb Rb Lss Rss
Sampling rate : 48.0 KHz
Frame rate : 93.750 fps (512 SPF)
Bit depth : 24 bit
Compression mode : Lossless
Stream size : 8.24 MiB (28%)
Title : Inverted Vuvuzela Test
Language : English
Default : Yes
Forced : No

Audio #2

ID : 3
Format : DTS
Format/Info : Digital Theater Systems
Codec ID : A_DTS
Duration : 15s 595ms
Bit rate mode : CBR
Bit rate : 1509 Kbps
Channel(s) : 6 channel
Channel layout : C L R Ls Rs LFE
Sampling rate : 48.0 KHz
Frame rate : 93.750 fps (512 SPF)
Compression mode : Lossy
Stream size : 2.81 MiB (10%)
Title : Extracted DTS Core
Language : English
Default : Yes
Forced : No

Audio #3

ID : 4
Format : E-AC-3
Format/Info : Enhanced AC-3
Commercial name : Dolby Digital Plus
Codec ID : A_EAC3
Duration : 15s 616ms
Bit rate mode : CBR
Bit rate : 1024 Kbps
Channel(s) : 8 channel
Channel layout : L R C LFE Ls Rs Lb Rb
Sampling rate : 48.0 KHz
Frame rate : 31.250 fps (1536 SPF)
Bit depth : 32 bit
Compression mode : Lossy
Stream size : 1.91 MiB (7%)
Title : EAC3 7.1 from DTS (Plex/EAE)
Language : English
Service kind : Complete Main
Default : Yes
Forced : No

Audio #4

ID : 5
Format : FLAC
Format/Info : Free Lossless Audio Codec
Codec ID : A_FLAC
Duration : 15s 595ms
Bit rate mode : VBR
Bit rate : 5416 Kbps
Channel(s) : 8 channel
Channel layout : L R C LFE Lb Rb Ls Rs
Sampling rate : 48.0 KHz
Frame rate : 11.735 fps (4090 SPF)
Bit depth : 24 bit
Compression mode : Lossless
Stream size : 10.1 MiB (35%)
Title : eac3to.exe DIDNEYVOOVINVF.dtshd DIDNEYVOOVINVF.dtshd_.flac -progressnumbers -log=UsEac3To.log
Writing library : libFLAC 1.2.1 (UTC 2007-09-17)
Language : English
Default : Yes
Forced : No

Audio #5

ID : 6
Format : E-AC-3
Format/Info : Enhanced AC-3
Commercial name : Dolby Digital Plus
Codec ID : A_EAC3
Duration : 15s 648ms
Bit rate mode : CBR
Bit rate : 1024 Kbps
Channel(s) : 8 channel
Channel layout : L R C LFE Ls Rs Lb Rb
Sampling rate : 48.0 KHz
Frame rate : 31.250 fps (1536 SPF)
Compression mode : Lossy
Stream size : 1.91 MiB (7%)
Title : eac3to -0,1,2,3,6,7,4,5 FLAC -> E-AC-3 (Adobe Audition 2015.2)
Language : English
Service kind : Complete Main
Default : Yes
Forced : No