eac3to - audio conversion tool

[tebasuna51]

eac3to + Nero Audio Decoder (Nero 7) are dropping first aac frame on decoding. always. along with faad. ffmpeg & Nero AAC Decoder 1.5.1.0 are doing it correctly.

I agree with the Nero 7 behaviour but not with faad (14/06/2010 from RareWares) and even with NeroAacDec 1.5.1.0.

Encode/Decoder Nero7 NeroDec Faad Qaac ffmpeg
--------------- ------ ------- ---- ---- ------
2.0-nero.m4a - Ok +33 Ok Ok
5.1-nero.m4a - Ok +33 Ok Ok
2.0-qaac.aac +22 - +22 +44 +44
2.0-qaac.m4a - Ok +22 Ok Ok
5.1-qaac.aac broken - +22 +44 +44
5.1-qaac.m4a - Ok +22 Ok Ok
2.0-qaac-ND.aac -21 - -21 Ok Ok
2.0-qaac-ND.m4a - 0 -21 Ok Ok
5.1-qaac-ND.aac broken - -21 Ok Ok
5.1-qaac-ND.m4a - 0 -21 Ok Ok

ND = qaac --no-delay parameter
-21 = first frame cutted
+x = ms of silence delay added
0 = first frame silenced
- = Not supported

[kukushka]

Did you also try neroAacDec alone to see if the first aac frame is dropped ?
"Nero AAC Decoder 1.5.1.0" - this is neroAacDec and it's all good with it.

I agree with the Nero 7 behaviour but not with faad (14/06/2010 from RareWares) and even with NeroAacDec 1.5.1.0.

the thing with faad is that it is broken even more, not only it drops the first frame, it also ignores initial m4a header that tells other decoders (that can work with m4a directly obviously) what to skip. and no silence is added by decoders (qaac --no-delay shows it clearly), it is coded in aac and then skipped or decoded with or without frame drop depending on container/decoder

tests are for 2.0


process-decoder\coder nero qaac default qaac --no-delay
samples ms samples ms samples ms
m4a-faad 1600 33,33 1088 22,67 -1024 -21,33
m4a-ffmpeg 0 0 0 0 0 0
aac-eac3to&faad 1600 33,33 1088 22,67 -1024 -21,33
aac-ffmpeg 2624 54,67 2112 44 0 0
aac-mp4-faad 1600 33,33 1088 22,67 -1024 -21,33
aac-mp4-ffmpeg 2624 54,67 2112 44 0 0
aac-mkv-eac3to 1600 33,33 1088 22,67 -1024 -21,33
aac-mkv-ffmpeg 2624 54,67 2112 44 0 0
mkv w delay-eac3to -1040 -21,67 -1024 -21,33
mkv w delay-ffmpeg -448 -9,333 -960 -20
mkv-eac3to -1472 -30,67 -1984 -41,33 -1024 -21,33
mkv-ffmpeg -448 -9,333 -960 -20 0 0
ts-eac3to 1600 33,33 1088 22,67 -1024 -21,33
ts-ffmpeg 2624 54,67 2112 44 0 0


i omitted some identical results (like neroaacdec behavior = ffmpeg), they're mentioned in conclusions
m4a - initial encode
aac - demux from m4a with mp4muxer
aac-mp4 - remuxed back with mp4muxer, initial headers got stripped
aac-mkv - mkvtoolnix 8.5.2 was used
mkv w delay - m4a to mkv with additional track - container delays, 20ms for default qaac, 9ms for nero
ts - muxed from m4a with tsmuxer 1.10.6
versions:
eac3to 3.30, nerodec 1.5.1.0, neroenc 1.5.4.0, qaac 2.55-7.10.5.0, ffmpeg N-76417-gee20354, faad 373248B
conclusions:
encoders: nero - 2624 samples initial delay, qaac - 2112 by default
faad don't give a damn about m4a header
faad & eac3to both kill first frame (1024samples)
nero & ffmpeg (and foobar) -ok in m4a
mkv muxing from m4a tries to compensate coder delays by killing a few frames. and then adding a delay. or not adding when it's a single track which leads to -960 samples with qaac when properly decoded or almost 2k with nero7 frame drop
foobar is using ffmpeg engine so results are identical
ffmpeg (with foobar) don't count mkv delays
neroaacdec & ffmpeg results are identical too

[tebasuna51]

@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.
EDIT: after more test the cuts are unpredictables (from 0 to 54 ms)

- 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).

[Overdrive80]

Hi, folks. I purchase bluray and get this errors with eac3to:

M2TS, 1 video track, 5 audio tracks, 6 subtitle tracks, 1:49:25, 28.878p
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, Spanish, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 768kbps, 48kHz)
4: DTS Master Audio, Spanish, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 768kbps, 48kHz)
5: DTS Master Audio, Spanish, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 768kbps, 48kHz)
6: DTS Master Audio, Spanish, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 768kbps, 48kHz)
7: DTS Master Audio, Spanish, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 768kbps, 48kHz)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
[a03] Extracting audio track number 3...
[a03] Decoding with libDcaDec DTS Decoder...
[a03] libDcaDec reported the warning "XLL output not lossless". <WARNING>
[a03] Writing WAV...
[a03] Creating file "E:\Bluray\DBZ BOG EXTENDED\_3spa.wav"...
[a03] Skipping identical DTS frames (seamless branching)...
[a03] libDcaDec output changed from 6 channels, 48kHz to 2 channels, 48kHz. <ERROR>
Aborted at file position 21188608. <ERROR>

I attempt decoding using Arcsoft but issue report is the same.

Command line: "eac3to.exe" "E:\Bluray\DBZ BOG EXTENDED\" 1) 3: "E:\Bluray\DBZ BOG EXTENDED\_3spa.wav" -progressnumbers

EDIT: If I use m2ts directly on eac3to, work fine. I think that is one problem of playlist.

EDIT2: I ever get audio 2 channels with mt2s, but not 5.1.

[Thunderbolt8]

did you check the playlist file of how many .m2ts files it consists (the file itself, not what eac3to displays)? perhaps the main .m2ts file is preceeded by some other small .m2ts file which has a 5.1 channel configuration.

[Overdrive80]

did you check the playlist file of how many .m2ts files it consists (the file itself, not what eac3to displays)? perhaps the main .m2ts file is preceeded by some other small .m2ts file which has a 5.1 channel configuration.

Yes, its composed of two m2ts (0003.mt2s+0000.m2ts), however 0003.m2ts is short video of presentation of distribution company, nothing of interest.

I can extract audio track in dtshd (1,3 gb), but is impossible transcoding, ever get error.

[Thunderbolt8]

then just skip that one and only work with the main movie file.

[Overdrive80]

then just skip that one and only work with the main movie file.

The problem is that main movie info 2 channels from m2ts, but dtshd extracted info 6 channels, and I can not decoding audio:

General
Nombre completo : E:\Bluray\DBZ BOG EXTENDED\00000.mpls_3spa.dtshd
Formato : DTS
Formato/Info : Digital Theater Systems
Formato del perfil : MA / Core
Tamaño del archivo : 1,23GIB
Modo de tasa de bits : Variable

Audio
Formato : DTS
Formato/Info : Digital Theater Systems
Formato del perfil : MA / Core
Format_Settings_Mode : 16
Ajustes del formato, Endianness : Big
Tipo de tasa de bits : Variable
Tasa de bits : Desconocido / 768Kbps
Canal(es) : 6canales
Posiciones del canal : Front: L C R, Side: L R, LFE
Velocidad de muestreo : 48,0KHz
BitDepth/String : 16bits
Compression_Mode/String : / Lossy

From m2ts:

General
ID : 0 (0x0)
Nombre completo : E:\Bluray\DBZ BOG EXTENDED\BDMV\STREAM\00000.m2ts
Formato : BDAV
Formato/Info : Blu-ray Video
Tamaño del archivo : 21,8GIB
Duración : 1h 49min.
Modo de tasa de bits : Variable
Tasa de bits total : 28,6Mbps
Tasa de bits máxima : 48,0Mbps

Video
ID : 4113 (0x1011)
ID Menú : 1 (0x1)
Formato : AVC
Formato/Info : Advanced Video Codec
Formato del perfil : High@L4.1
Ajustes del formato, CABAC : Si
Ajustes del formato, RefFrames : 4marcos
ID Códec : 27
Duración : 1h 49min.
Tipo de tasa de bits : Variable
Tasa de bits máxima : 35,0Mbps
Ancho : 1 920pixeles
Alto : 1 080pixeles
Relación de aspecto : 16:9
Velocidad de cuadro : 23,976fps
Estándar : NTSC
ColorSpace : YUV
ChromaSubsampling : 4:2:0
BitDepth/String : 8bits
Tipo de exploración : Progresivo
colour_range : Limited
colour_primaries : BT.709
transfer_characteristics : BT.709
matrix_coefficients : BT.709

Audio #1
ID : 4352 (0x1100)
ID Menú : 1 (0x1)
Formato : DTS
Formato/Info : Digital Theater Systems
Formato del perfil : MA / Core
Format_Settings_Mode : 16
Ajustes del formato, Endianness : Big
Modo Muxing : Stream extension
ID Códec : 134
Duración : 1h 49min.
Tipo de tasa de bits : Variable
Tasa de bits : Desconocido / 1 509Kbps
Canal(es) : 2canales
Posiciones del canal : Front: L R
Velocidad de muestreo : 48,0KHz
BitDepth/String : 16bits
Compression_Mode/String : / Lossy

[SeeMoreDigital]

Over the last few months I've been backing-up all my multi-channel (DVD-A, Blu-Ray Audio and DTS-CD) albums to single multi-channel flac files complete with a cue file.

All had been going very well up-until this morning, when I tried to back-up my 'Don Henley - End Of The Innocence' DTS-CD. After some investigation I've determined that the problem lies with the disc's very first track, which has been encoded using 6.1 DTS-ES audio, instead of regular 5.1 DTS audio.

So far I've managed to back-up the first track to a (52.6MB) dts-es.wav file. And I've also managed to extract the elementary (46.0MB) dts-es.dts stream - but now I'm stumped :eek:

Ideally what I'd like to do is create a 5.1 channel flac file from the 6.1 channel dts-es source but sadly neither LameXP or UsEac3to (with eac3to v3.31) are able to do this.

Does anyone have any ideas?

EDIT: Here's a link to a 5 second DTS-ES sample (https://www.sendspace.com/file/lj6nys)

[Thunderbolt8]

The problem is that main movie info 2 channels from m2ts, but dtshd extracted info 6 channels, and I can not decoding audio:

General
Nombre completo : E:\Bluray\DBZ BOG EXTENDED\00000.mpls_3spa.dtshd
Formato : DTS
Formato/Info : Digital Theater Systems
Formato del perfil : MA / Core
Tamaño del archivo : 1,23GIB
Modo de tasa de bits : Variable

Audio
Formato : DTS
Formato/Info : Digital Theater Systems
Formato del perfil : MA / Core
Format_Settings_Mode : 16
Ajustes del formato, Endianness : Big
Tipo de tasa de bits : Variable
Tasa de bits : Desconocido / 768Kbps
Canal(es) : 6canales
Posiciones del canal : Front: L C R, Side: L R, LFE
Velocidad de muestreo : 48,0KHz
BitDepth/String : 16bits
Compression_Mode/String : / Lossy

From m2ts:

General
ID : 0 (0x0)
Nombre completo : E:\Bluray\DBZ BOG EXTENDED\BDMV\STREAM\00000.m2ts
Formato : BDAV
Formato/Info : Blu-ray Video
Tamaño del archivo : 21,8GIB
Duración : 1h 49min.
Modo de tasa de bits : Variable
Tasa de bits total : 28,6Mbps
Tasa de bits máxima : 48,0Mbps

Video
ID : 4113 (0x1011)
ID Menú : 1 (0x1)
Formato : AVC
Formato/Info : Advanced Video Codec
Formato del perfil : High@L4.1
Ajustes del formato, CABAC : Si
Ajustes del formato, RefFrames : 4marcos
ID Códec : 27
Duración : 1h 49min.
Tipo de tasa de bits : Variable
Tasa de bits máxima : 35,0Mbps
Ancho : 1 920pixeles
Alto : 1 080pixeles
Relación de aspecto : 16:9
Velocidad de cuadro : 23,976fps
Estándar : NTSC
ColorSpace : YUV
ChromaSubsampling : 4:2:0
BitDepth/String : 8bits
Tipo de exploración : Progresivo
colour_range : Limited
colour_primaries : BT.709
transfer_characteristics : BT.709
matrix_coefficients : BT.709

Audio #1
ID : 4352 (0x1100)
ID Menú : 1 (0x1)
Formato : DTS
Formato/Info : Digital Theater Systems
Formato del perfil : MA / Core
Format_Settings_Mode : 16
Ajustes del formato, Endianness : Big
Modo Muxing : Stream extension
ID Códec : 134
Duración : 1h 49min.
Tipo de tasa de bits : Variable
Tasa de bits : Desconocido / 1 509Kbps
Canal(es) : 2canales
Posiciones del canal : Front: L R
Velocidad de muestreo : 48,0KHz
BitDepth/String : 16bits
Compression_Mode/String : / Lossydid you really use the 00000.m2ts file to extract the audio from and not the .mpls file? because according to the file name of your .dtshd track this is at least what is indicated here.

[Overdrive80]

Yes, I used playlist and mt2s. From playlist can only extract dts files, with m2ts can extract and transcoding but I never get audio 5.1 only 2.0. Is posible that this company had hack playlist for showing false dts 5.1?

[ndjamena]

Where is it showing false information?

The first m3ts contains 5.1 audio, the second 2.0. If you extract the playlist the header at the beginning will come from the first file and show as 5.1. If you only extract from the second m2ts it will always show as 2.0, because that is what is in the file.

There is no 5.1 to decode, unless you like the intro in the first m2ts enough that you'd like to keep it.

Your options are to ditch the intro, or downmix the sound from it to stereo and prepend it to the audio from the second m2ts file.

[Overdrive80]

And can not it done intentionally?? Company sold product like DTS-MA 5.1: http://www.amazon.es/Dragon-Battle-Edic-Coleccionista-Blu-ray/dp/B00VVQD7YI/ref=pd_sim_74_6?ie=UTF8&dpID=6127ukb533L&dpSrc=sims&preST=_AC_UL160_SR158%2C160_&refRID=1A7QSJVR19TT8QQ27FWS (Pre-sold that I got)

I am refering to false info to:
http://s7.postimg.org/o739b1dg7/Captura_de_pantalla_2015_11_22_17_40_16.jpg (http://postimg.org/image/o739b1dg7/)

[Thunderbolt8]

in that case you could try to contact the company and tell them about it & complain, get a refund etc.

[tebasuna51]

Ideally what I'd like to do is create a 5.1 channel flac file from the 6.1 channel dts-es source but sadly neither LameXP or UsEac3to (with eac3to v3.31) are able to do this.

Without problem here using your sample and:

%_.flac -down6

eac3to v3.31
command line: "D:\Programa\eac3to\eac3to.exe" "D:\tmp\5 sec DTS-ES elementary stream sample.dts" "D:\tmp\5 sec DTS-ES elementary stream sample.dts_.flac" -down6 -progressnumbers -log="D:\Programa\eac3to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
DTS-ES, 6.1 channels, 0:00:05, 1235kbps, 44.1kHz
Decoding with libDcaDec DTS Decoder...
Mixing surround channels...
Encoding FLAC with libFlac...
Creating file "D:\tmp\5 sec DTS-ES elementary stream sample.dts_.flac"...
eac3to processing took 1 second.
Done.

FLAC, 5.1 channels, 0:00:05, 24 bits, 3530kbps, 44.1kHz
Also ok to wav. I don't know how you obtain your "5 sec DTS-ES in WAV sample.wav"
WAV, 2.0 channels, 0:00:05, 16 bits, 1411kbps, 44.1kHz

[SeeMoreDigital]

Without problem here using your sample and:
%_.flac -down6


I actually tried these command line parameters with UsEac3To but I get this warning: -

eac3to v3.31
command line: "C:\UsEac3to118\eac3to.exe" "C:\Users\SeeMoreDigital\Desktop\The End of the Innocence.dts" "C:\Users\SeeMoreDigital\Desktop\The End of the Innocence.dts_.flac" -down6 -progressnumbers -log="C:\UsEac3to118\UsEac3To.log"
------------------------------------------------------------------------------
DTS-ES, 6.1 channels, 0:05:13, 1235kbps, 44.1kHz
Decoding with libDcaDec DTS Decoder...
Mixing surround channels...
Encoding FLAC with libFlac...
Creating file "C:\Users\SeeMoreDigital\Desktop\The End of the Innocence.dts_.flac"...
libDcaDec reported the warning "Failed to parse core extension". <WARNING>
libDcaDec output changed from 7 channels, 44kHz to 6 channels, 44kHz. <ERROR>
Aborted at file position 1310720. <ERROR>

Cheers

[tebasuna51]

libDcaDec output changed from 7 channels, 44kHz to 6 channels, 44kHz. <ERROR>
Aborted at file position 1310720. <ERROR>
Yep, seems the problem you say: "the problem lies with the disc's very first track, which has been encoded using 6.1 DTS-ES audio, instead of regular 5.1"

Then you need split your dts in 2 parts, only the first one need the -down6

eac3to 1.dts 1.wav -down6
eac3to 2.dts 2.wav

After you can concatenate the 2 wav's:

sox 1.wav 2.wav full.wav

And encode to flac:

eac3to full.wav full.flac

The problem now is split the dts. Like "Aborted at file position 1310720", maybe you can upload me 2MB and I can say you how split the dts:

eac3to full.dts begin.dts -2mb

[SeeMoreDigital]

Yep, seems the problem you say: "the problem lies with the disc's very first track, which has been encoded using 6.1 DTS-ES audio, instead of regular 5.1"

maybe you can upload me 2MB and I can say you how split the dts...


Many thanks for looking into this. You're right the 5 second sample encodes fine but I've just cut off a 10 second sample which does not.

Here's a link to a 10 second DTS-ES sample (https://www.sendspace.com/file/vvxuft). And another to the full dts-es file (https://www.sendspace.com/file/c5hmpz).


Cheers

[tebasuna51]

@SeeMoreDigital

Is not so easy, the dts is corrupt and there are 37 frames along the full dts (13460 frames) with the BC channel unrecoverable.

eac3to crash decoding with dcadec and Arcsoft. With libav produce 346 WARNINGS and the decoded wav is useless.

I have succes with BeHappy decoding the dts to wav 6.1 with:
LWLibavAudioSource("D:\tmp\input.dts", stream_index=0,cache=false)
than use dcadec also. Here I see the 37 broken frames decoded as silence.

Using Foobar2000 I get a wav 5.1 with the BC channel already mixed in BL-BR. Sound fine for me: https://www.sendspace.com/file/lw3fl1

[SeeMoreDigital]

@SeeMoreDigital

Is not so easy, the dts is corrupt and there are 37 frames along the full dts (13460 frames) with the BC channel unrecoverable.

Thank-you so much for doing this.

So the actual dts-es elementary stream is at fault. Do you think this could this have happened when it was extracted from the .wav container using DTSParser v2. Or somewhere else?

Thanks again :)

[tebasuna51]

Do you think this could this have happened when it was extracted from the .wav container using DTSParser v2. Or somewhere else?

The dts play fine when I send it by spdif to my 5.1 receiver, maybe with a 6.1 or 7.1 we can listen some clicks in back speakers, but downmixed to 5.1 I can't listen anything strange.

Maybe the original DTS was wrong encoded or the original CD was dirty when ripped to wav, I don't think than the DTSParser was the culprit.
Try to clean the CD and rerip the track.

[SeeMoreDigital]

Maybe the original DTS was wrong encoded or the original CD was dirty when ripped to wav, I don't think than the DTSParser was the culprit.
Try to clean the CD and rerip the track.Update: I've cleaned the disc. And used several CD ripping applications, including EAC. But it would seem that the actual data for that particular track is corrupt in some way.

As a side note. Here's an interesting thing. If you encode the dts.wav file to flac. And then the flac file back to wav. The dts stream is still completely intact...


Cheers

[Soulvomit]

I wanted to get an 11:08.192 clip from a 1:52:59.136 AC3 from the 1:39:36.770 mark and after much trial and error I stumbled upon the combination of "-edit=1:50:44.939,-5976770ms" and "-5976770ms" to do so. With "-edit=1:50:44.939,-5976770ms" alone, the AC3 would decode up to 1:50:44.939. With "-5976770ms" alone, it would do so from 1:39:36.770 to the end, 1:52:59.136. Now is there a particular reason why trimming a track requires two options, in different formats no less?eac3to %%a.ac3 %%~na.wav -edit=1:50:44.939,-5976770ms -5976770ms -full -no2ndpass Why isn't such a pedestrian task simple and intuitive? Can a better, more uniform option for this be implemented, preferably in milliseconds, or better yet, samples? Something like:-ms-in=5976770 -ms-out=6644939 or -sample-in=286884998 -sample-out=318957039

[ndjamena]

Update: I've cleaned the disc. And used several CD ripping applications, including EAC. But it would seem that the actual data for that particular track is corrupt in some way.

As a side note. Here's an interesting thing. If you encode the dts.wav file to flac. And then the flac file back to wav. The dts stream is still completely intact...


Cheers

What was the bit rate of the FLAC?

[SeeMoreDigital]

What was the bit rate of the FLAC?
1,260Kbps (using compression level 6)...

[Thunderbolt8]

Madshi, could you please update dcadec in eac3to? The XXL not lossless warning seems to have been removed and as of November 27th it has reached version 0.1

[nevcairiel]

Madshi, could you please update dcadec in eac3to? The XXL not lossless warning seems to have been removed and as of November 27th it has reached version 0.1

The warning going away is just cosmetic though, so don't worry too much about it.

[madshi]

Furthermore you can just compile a new libdcadec.dll yourself and use it with eac3to. I'm not using any patches, so any newer version than the one I'm using should work fine.

[airsoft]

As we talk about DTS what about DTS:X support, at least a detection

Sent from my HTC Desire 516 dual sim using Tapatalk

[Thunderbolt8]

Furthermore you can just compile a new libdcadec.dll yourself and use it with eac3to. I'm not using any patches, so any newer version than the one I'm using should work fine.does it matter whether I compile and use the x64 or win32 one?

[Boulder]

Eac3to is a 32-bit application so it can only call x86 libraries.

[Thunderbolt8]

compared to a HDD, would running eac3to with 2 SSDs in raid mode 0 speed up remuxing blu-rays quite a lot? (reading and writing from the disk)

[heerschop]

Furthermore you can just compile a new libdcadec.dll yourself and use it with eac3to. I'm not using any patches, so any newer version than the one I'm using should work fine.

I now this is a bit of topic but I am trying to build the libdcadec.dll with visual studio 2015.
I succeeded in building the libdcadec.dll but the .dll is painfully slow in decoding my dts steams.
The libdcadec.dll that came with eac3to took about 2 minutes to decode the dts.
With my own build it took about 9 minutes to decode the same dts.
Are there any options in visual studio to set that influence the performance of the created libdcadec.dll?

[Groucho2004]

Are there any options in visual studio to set that influence the performance of the created libdcadec.dll?
Did you build a debug or a release DLL?

[heerschop]

Did you build a debug or a release DLL?

I did build a release dll.

[Groucho2004]

I did build a release dll.
In that case - no idea why it's so much slower. Try building it with GCC.

[Q-the-STORM]

madshi, do you have any intention of including speedup/slowdown functionality for chapters? As far as I can see there are no chapter programs that let you speedup via CLI, so speeding up multiple files is a big hassle... since it is just a simple calculation, it should not be that hard to implement (I guess)... though I know chapter editing is not really eac3to's purpose, so I get it if you don't implement it...

on another matter, I'm gonna dig up a question I asked almost 2 years ago:
Are you going to replace libaften with libav for ac3 encoding in the foreseeable future? last time I asked, you said you haven't gotten around to it... since libav is the best choice for ac3 encoding (apart from the official dolby pro encoder, which I guess can't be implemented in eac3to :P) it would be nice to have it directly in eac3to and not have to pipe it to ffmpeg every time...

[tebasuna51]

madshi, do you have any intention of including speedup/slowdown functionality for chapters?

By the moment you can try the GUI UsEac3to:

- Load the .txt (Chapters OGM) or .xml chapters in UsEac3to.
Also text subs .srt, .ssa or .ass are supported.

- Ignore the eac3to message:
"The format of the source file could not be detected. <ERROR>"

- Click in the 'Auxiliary tools' -> 'SRT/.../TXT'

- Select your desired conversion and 'Convert'

[Q-the-STORM]

By the moment you can try the GUI UsEac3to:

this seems to work for single files only... There are already a few programs out there that do that, I need it to convert multiple files at once, which is only possible if the either the GUI is able to read and convert multiple source files, or if there is a CLI, where I can use a for loop in cmd to process all files in one go...

[tebasuna51]

this seems to work for single files only...

I can add a option to process all files with the same extension in a folder.
Wait to a new version in UsEac3to thread (http://forum.doom9.org/showthread.php?t=145574).

EDIT: done

[Boulder]

Is it possible to output the file created during the first pass to some different location than where the actual destination file will be written to? It would make things quite a bit faster if I could output it to a different HD.

[arrgh]

madshi, do you have any intention of including speedup/slowdown functionality for chapters? As far as I can see there are no chapter programs that let you speedup via CLI, so speeding up multiple files is a big hassle...

try ChapterGen.exe... it is old and a Little bit bitchy, but works in CLI....

[Thunderbolt8]

is eac3to actually multithreaded, does it use multiple cpu cores? I wonder how remux/demux speed would be affected in the not too distant future when PCIe drives hit the 1000MB/s write speed wall.

[heerschop]

Try building it with GCC.

Thank you for your suggestion. I tried to compile with gcc (mingw) but no luck. I kept keep getting "undefined reference" errors.
I am not a developer so I don't know where to begin to fix these errors.
After fiddling about with visual studio and gcc with no success I give up on building the dcadec.lib.

thanks

[Boulder]

In Code Generation, you could try enabling the Enhanced Instruction Set option according to your CPU.

[heerschop]

In Code Generation, you could try enabling the Enhanced Instruction Set option according to your CPU.

In visual studio I have build the libdcadec.dll a zillion times with all sort of settings enabled or disabled. No matter what I did, all my builds were very slow in decoding a dts stream.
I even tried different visual studio versions (2010,2012,2013,2015).

[Boulder]

Weird..I've just tested with VS2015 and the DLL it creates, doesn't appear to be slow. I didn't make any specific comparisons but looking at the eac3to progress meter, I couldn't tell if the DLL was updated or not.

I have a question of my own: Moonrise Kingdom has a DTS-HD MA track which shows up as 5.0ch but actually is 5.1ch. Decoding with eac3to doesn't proceed because of that, and dcadec.exe reports "Error writing WAV file: PCM output parameters changed". Is there a way to patch the header to make it work?

[nevcairiel]

I have a question of my own: Moonrise Kingdom has a DTS-HD MA track which shows up as 5.0ch but actually is 5.1ch. Decoding with eac3to doesn't proceed because of that, and dcadec.exe reports "Error writing WAV file: PCM output parameters changed". Is there a way to patch the header to make it work?

It probably changes from 5.0 to 5.1ch somewhere, and if dcadec and eac3to don't handle such changes, there isn't much that can be done.

You could try instructing it to mix to 5.1 always, maybe that helps, not sure.

[heerschop]

Weird..I've just tested with VS2015 and the DLL it creates, doesn't appear to be slow.

I am new to building with visual studio. This is how I made the build.
- First download and installed Visual studio 2015
- Downloaded the dcadec source files => dcadec-master.zip (from https://github.com/foo86/dcadec )
- extracted the dcadec-master.zip
- opened the dcadec.sln in visual studio
- visual studio asks if I want update the solution to version 140 => Yes
- Build the libdcadec as a release dll. ( The resulting build is smaller in size then the one that came with eac3to)
- Copy the libdcadec.dll to my eac3to folder

Eac3to uses my build without a problem but the decoding is very slow.
Am I missing something or do I use the wrong source files?

[Boulder]

It seems to change almost immediately because the error message kicks in right after the decoding starts. The -down6 parameter doesn't help here either :(