eac3to - audio conversion tool - Page 742

zeropc · 6th May 2019, 01:32

Guys, can eac3to no longer slowdown videos in h265?
I get the following error message

Code:

C:\Users\zeropc>"\\Mac\Home\Desktop\eac3to more gui\eac3to\eac3to.exe" "X:\evil2mkv"
 1: "\\Mac\Home\Desktop\dead2\hdr.h264" 2: "\\Mac\Home\Desktop\dead2\dv.h264" -slowdown
MKV, 2 video tracks, 2 audio tracks, 1 subtitle track, 1:24:32, 24p
1: h265/HEVC, 2160p24 (16:9), 10 bits
2: h265/HEVC, 1080p24 (16:9), 10 bits - Dolby Vision Enhancement Layer
3: DTS Master Audio, English, 1.0 channels, 16 bits, 48kHz
   (core: DTS, 1.0 channels, 768kbps, 48kHz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
   (core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: Subtitle (PGS), English
v01 The video framerate is correct, but rather unusual.
v02 The video framerate is correct, but rather unusual.
This video conversion is not supported.

tebasuna51 · 6th May 2019, 09:41

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."

zeropc · 6th May 2019, 13:36

Quote:

Originally Posted by tebasuna51

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."

AHHH!!!

Would be sweet if madshi can add this option for h265

SeeMoreDigital · 6th May 2019, 13:50

Quote:

Originally Posted by tebasuna51

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."

Out of interest... Has anybody created a tool that can change some of the attributes of h265 elementary streams?

nevcairiel · 6th May 2019, 14:01

Quote:

Originally Posted by SeeMoreDigital

Out of interest... Has anybody created a tool that can change some of the attributes of h265 elementary streams?

Depending on what partciular properties you are talking about, the h265_metadata bitstream filter in ffmpeg can change a bunch of them.

sneaker_ger · 6th May 2019, 14:30

* hevc_metadata

SeeMoreDigital · 6th May 2019, 14:37

Marvellous...

Thank-you Mr sneaker or is it Mr ger

nevcairiel · 6th May 2019, 14:37

How misleading, the file was called h265_metadata_bsf.c

an3k · 16th May 2019, 19:37

I noticed a bug with eac3to v3.34 and Surcode DTS Encoder. When you convert a multichannel AC3 or WAV (for FLAC, etc.) to DTS it works fine. If you however want to convert a singlechannel (1.0 aka. mono) to DTS eac3to - or better Surcode throws an error. The very same file eac3to doesn't handle correctly is perfectly working if used directly inside the Surcode DTS Encoder software.

The error is:

Quote:

C:\Program Files (x86)\MeGUI\tools\eac3to>eac3to.exe "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.wav" "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.dts"
WAV, 1.0 channels, 2:32:13, 24 bits, 1152kbps, 48kHz
Reading WAV...
Writing WAVs...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
The original audio track has a constant bit depth of 24 bits.
Encoding DTS <768kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Surcode says/asks: "Invalid Wave File D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.dts.".
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Somehow eac3to pushes the destination filename to Surcode as the source filename.

Another bug is that if your source file is named eg. "foobar.C.wav" and you want to convert it to AC3 or DTS eac3to wants to convert that file to "foobar.C.wav" to make it compatible with the format AC3 or DTS expects. However, it throws an error because it just overwrote the source audio file with the temporary one

tebasuna51 · 16th May 2019, 23:28

@an3k
Encode a monowav to standard DTS 768 Kb/s with Surcode is usseless, you obtain a file with the same size and less quality than the PCM 16 bits (WAV).

eac3to don't need manage that conversion.

I can't reproduce the name problem encoding to AC3.

an3k · 16th May 2019, 23:48

Quote:

Originally Posted by tebasuna51

@an3k
Encode a monowav to standard DTS 768 Kb/s with Surcode is usseless, you obtain a file with the same size and less quality than the PCM 16 bits (WAV).

eac3to don't need manage that conversion.

I can't reproduce the name problem encoding to AC3.

The sense may be non-existend. I agree with you on that point but non-the-less it is a bug in eac3to.

Using the same file directly in Surcode works (even though the resulting DTS file is 5.1 channels). Because of that I went and created a DTS-HD MA using the DTS Master Audio Suite which created a mono file.

Yeah, if the target is AC3 it works without issues.

an3k · 19th May 2019, 12:54

I noticed another weird behavior and don't know if eac3to is doing something wrong or me not understanding it.

Code:

eac3to v3.34
command line: eac3to.exe  "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.ac3" "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.wavs"
------------------------------------------------------------------------------
AC3, 5.1 channels, 2:32:13, 448kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.L.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SL.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SR.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.LFE.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.R.wav"...
Clipping detected, a 2nd pass will be necessary.  <WARNING>
Starting 2nd pass...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Applying -0.55dB gain...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.L.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.LFE.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.R.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SL.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SR.wav"...
eac3to processing took 5 minutes, 17 seconds.
Done.

eac3to says

Code:

Reducing depth from 64 to 24 bits...

but MediaInfo says the ac3 is 16 bits. The second audio stream is TrueHD (MediaInfo says it's 16 bits too) and if I transcode that into individual wavs eac3to correctly use 16 bits.

sneaker_ger · 19th May 2019, 14:01

AC3 and other lossy codecs don't have any inherent bitdepth despite what MediaInfo might suggest. The decoder can decode at 64 bits but it would be to wasteful to store 64 bit PCM/WAV so eac3to reduces the depth.

an3k · 19th May 2019, 14:35

Oh, I see. Thank you for the information.

What about Dolby TrueHD? MediaInfo says 24 bits but eac3to says 16 bits and is reducing from 24 to 16 bits.

I want to keep as much quality as possible and the size of the intermediate wav files isn't a thing I have to worry about since I'm going to encode to a different format.

sneaker_ger · 19th May 2019, 14:52

Lossless codecs do have "inherent" bitdepths. In your case it was encoded as 24 bits. But eac3to detects that throughout the complete file all of the respective 8 minor bits are 0s. So it decides to encode to 16 bit instead. This has absolutely no influence on quality.

MeteorRain · 24th May 2019, 16:30

Is there any way to stop eac3to from adding DELAY to output file name?

I'm writing script to batch extract eac3 from files and convert them to wav for further processing.

eac3to x.mkv x.wav tells me This audio conversion is not supported..

eac3to x.mkv x.eac3 then eac3to x.eac3 x.wav works.
However eac3to will add DELAY XYZ ms to the file name so it's kinda hard to script that into a makefile.
Since this delay is usually less than 1 frame, I really don't care if it's applied correctly.

sneaker_ger · 24th May 2019, 21:13

I would recommend using ffmpeg. It seems better at reading from mkv and also has a more feature-complete eac3 decoder IIRC.

Code:

ffmpeg -i "x.mkv" -map 0:a:0 "x.wav"

It can correct delays, too. Hope this is correct:

Code:

ffmpeg -i "x.mkv" -map 0:a:0 -af aresample=first_pts=0 "x.wav"

MeteorRain · 25th May 2019, 05:56

Thanks. I'm aware of ffmpeg but still prefer eac3to if possible.

thecoreyburton · 13th June 2019, 23:17

I often use eac3to to demux BD streams. In the past I've noted a common delay of just above 21ms (there's a remaining delay of less than 1ms) when dealing with DTSHD track transcoding (eg. DTSHD > FLAC). At the time, I read up on this a fair bit and did some of my own testing, eventually concluding that a simple "-21ms" parameter would be ideal when dealing with DTSHD in this way.

There's plenty of discussion about this, but a lot of it seems related to specific issues and I can't find anything conclusive about whether or not this is the right way to approach this. I'm not sure if there is a "correct" way to do this (for instance a trim based on samples / frames to make it align perfectly or some internal function that does this), or if I should be touch this value at all. I believe it's an encoder delay (and not a decoder delay, but I could be wrong) and so it stands to reason that if this is true, perhaps some BDs have been authored with compensation for this delay in mind.

I was wondering if the way I'm handling it is the correct procedure, or is there a better way?

tebasuna51 · 14th June 2019, 02:33

It is know than encoders add initial silence samples to initialize some encoders buffers, for instance if you recode to AC3 you obtain a delay of 5 ms, until 38 ms with some AAC encoders.

But with some encoders that delay is stored in metadata and any decoder must restore the audio without delay. For instance AAC in mp4 container (.m4a) must play without delay.
I can't reproduce the problem recoding DTSHD to FLAC, for me the obtained .flac is decoded without delay.

6th May 2019, 09:41	#14822 \| Link
tebasuna51 Moderator Join Date: Feb 2005 Location: Spain Posts: 6,009	eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is: "This video conversion is not supported." __________________ BeHappy, AviSynth audio transcoder.

6th May 2019, 14:37	#14827 \| Link
SeeMoreDigital Life looks better in UHD Join Date: Jun 2003 Location: Notts, UK Posts: 11,724	Marvellous... Thank-you Mr sneaker or is it Mr ger __________________ \| I've been testing hardware media playback devices and software A/V encoders and decoders since 2001 \| My Network Layout & A/V Gear \|

6th May 2019, 14:37	#14828 \| Link
nevcairiel Registered Developer Join Date: Mar 2010 Location: Hamburg/Germany Posts: 10,027	How misleading, the file was called h265_metadata_bsf.c __________________ LAV Filters - open source ffmpeg based media splitter and decoders

16th May 2019, 23:28	#14830 \| Link
tebasuna51 Moderator Join Date: Feb 2005 Location: Spain Posts: 6,009	@an3k Encode a monowav to standard DTS 768 Kb/s with Surcode is usseless, you obtain a file with the same size and less quality than the PCM 16 bits (WAV). eac3to don't need manage that conversion. I can't reproduce the name problem encoding to AC3. __________________ BeHappy, AviSynth audio transcoder.

24th May 2019, 16:30	#14836 \| Link
MeteorRain 結城有紀 Join Date: Dec 2003 Location: NJ; OR; Shanghai Posts: 871	Is there any way to stop eac3to from adding DELAY to output file name? I'm writing script to batch extract eac3 from files and convert them to wav for further processing. eac3to x.mkv x.wav tells me This audio conversion is not supported.. eac3to x.mkv x.eac3 then eac3to x.eac3 x.wav works. However eac3to will add DELAY XYZ ms to the file name so it's kinda hard to script that into a makefile. Since this delay is usually less than 1 frame, I really don't care if it's applied correctly. __________________ Projects x265 - Yuuki-Asuna-mod Download / GitHub TS - ADTS AAC Splitter \| LATM AAC Splitter \| BS4K-ASS Neo AviSynth+ filters - F3KDB \| FFT3D \| DFTTest \| MiniDeen \| Temporal Median

6th May 2019, 14:30	#14826 \| Link
sneaker_ger Registered User Join Date: Dec 2002 Posts: 5,563	* hevc_metadata

19th May 2019, 14:01	#14833 \| Link
sneaker_ger Registered User Join Date: Dec 2002 Posts: 5,563	AC3 and other lossy codecs don't have any inherent bitdepth despite what MediaInfo might suggest. The decoder can decode at 64 bits but it would be to wasteful to store 64 bit PCM/WAV so eac3to reduces the depth.

19th May 2019, 14:35	#14834 \| Link
an3k Registered User Join Date: Oct 2006 Location: Munich, Germany Posts: 171	Oh, I see. Thank you for the information. What about Dolby TrueHD? MediaInfo says 24 bits but eac3to says 16 bits and is reducing from 24 to 16 bits. I want to keep as much quality as possible and the size of the intermediate wav files isn't a thing I have to worry about since I'm going to encode to a different format.

19th May 2019, 14:52	#14835 \| Link
sneaker_ger Registered User Join Date: Dec 2002 Posts: 5,563	Lossless codecs do have "inherent" bitdepths. In your case it was encoded as 24 bits. But eac3to detects that throughout the complete file all of the respective 8 minor bits are 0s. So it decides to encode to 16 bit instead. This has absolutely no influence on quality.

24th May 2019, 21:13	#14837 \| Link
sneaker_ger Registered User Join Date: Dec 2002 Posts: 5,563	I would recommend using ffmpeg. It seems better at reading from mkv and also has a more feature-complete eac3 decoder IIRC. Code: ffmpeg -i "x.mkv" -map 0:a:0 "x.wav" It can correct delays, too. Hope this is correct: Code: ffmpeg -i "x.mkv" -map 0:a:0 -af aresample=first_pts=0 "x.wav"

25th May 2019, 05:56	#14838 \| Link
MeteorRain 結城有紀 Join Date: Dec 2003 Location: NJ; OR; Shanghai Posts: 871	Thanks. I'm aware of ffmpeg but still prefer eac3to if possible. __________________ Projects x265 - Yuuki-Asuna-mod Download / GitHub TS - ADTS AAC Splitter \| LATM AAC Splitter \| BS4K-ASS Neo AviSynth+ filters - F3KDB \| FFT3D \| DFTTest \| MiniDeen \| Temporal Median

13th June 2019, 23:17	#14839 \| Link
thecoreyburton Registered User Join Date: Jul 2015 Posts: 104	I often use eac3to to demux BD streams. In the past I've noted a common delay of just above 21ms (there's a remaining delay of less than 1ms) when dealing with DTSHD track transcoding (eg. DTSHD > FLAC). At the time, I read up on this a fair bit and did some of my own testing, eventually concluding that a simple "-21ms" parameter would be ideal when dealing with DTSHD in this way. There's plenty of discussion about this, but a lot of it seems related to specific issues and I can't find anything conclusive about whether or not this is the right way to approach this. I'm not sure if there is a "correct" way to do this (for instance a trim based on samples / frames to make it align perfectly or some internal function that does this), or if I should be touch this value at all. I believe it's an encoder delay (and not a decoder delay, but I could be wrong) and so it stands to reason that if this is true, perhaps some BDs have been authored with compensation for this delay in mind. I was wondering if the way I'm handling it is the correct procedure, or is there a better way?

14th June 2019, 02:33	#14840 \| Link
tebasuna51 Moderator Join Date: Feb 2005 Location: Spain Posts: 6,009	It is know than encoders add initial silence samples to initialize some encoders buffers, for instance if you recode to AC3 you obtain a delay of 5 ms, until 38 ms with some AAC encoders. But with some encoders that delay is stored in metadata and any decoder must restore the audio without delay. For instance AAC in mp4 container (.m4a) must play without delay. I can't reproduce the problem recoding DTSHD to FLAC, for me the obtained .flac is decoded without delay. __________________ BeHappy, AviSynth audio transcoder.

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