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#14821 | Link | |
Moderator
![]() Join Date: Feb 2005
Location: Spain
Posts: 5,660
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Quote:
The best AC3 free encoder (ffmpeg) used with MeGUI or BeHappy is not a Dolby Digital encoder. The recommended Dialog Normalization and Dynamic Range Compression with this encoder is the default (-31 dB and None), then is not necesary know the RMS value of center channel in Sound Forge (or with other method). Take in mind than you are recoding a previous track, not encoding material recorded by you. Many encoder options in commercial encoders (filters, phase shift, etc.) are now unnecesary.
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BeHappy, AviSynth audio transcoder, in Doom9 forums. NicAudio, BassAudio, audio decoders. |
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#14822 | Link | |||
Registered User
Join Date: May 2017
Posts: 2
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Quote:
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And thank you very much for the help, I read more half of the pages of this topic and I observed that you are a helpful person. Greetings from Brazil. |
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#14823 | Link |
Registered User
Join Date: Mar 2007
Posts: 100
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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. Last edited by tebasuna51; 6th May 2019 at 10:23. |
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#14824 | Link |
Moderator
![]() Join Date: Feb 2005
Location: Spain
Posts: 5,660
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eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."
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BeHappy, AviSynth audio transcoder, in Doom9 forums. NicAudio, BassAudio, audio decoders. |
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#14826 | Link |
Life looks better in UHD
Join Date: Jun 2003
Location: Notts, UK
Posts: 11,455
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Out of interest... Has anybody created a tool that can change some of the attributes of h265 elementary streams?
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| I've been testing hardware media playback devices and software A/V encoders and decoders since 2001 | My Network Layout & A/V Gear |
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#14827 | Link |
Registered Developer
Join Date: Mar 2010
Location: Hamburg/Germany
Posts: 9,833
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Depending on what partciular properties you are talking about, the h265_metadata bitstream filter in ffmpeg can change a bunch of them.
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LAV Filters - open source ffmpeg based media splitter and decoders |
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#14829 | Link |
Life looks better in UHD
Join Date: Jun 2003
Location: Notts, UK
Posts: 11,455
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Marvellous...
Thank-you Mr sneaker or is it Mr ger ![]()
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| I've been testing hardware media playback devices and software A/V encoders and decoders since 2001 | My Network Layout & A/V Gear |
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#14830 | Link |
Registered Developer
Join Date: Mar 2010
Location: Hamburg/Germany
Posts: 9,833
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How misleading, the file was called h265_metadata_bsf.c
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LAV Filters - open source ffmpeg based media splitter and decoders |
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#14831 | Link | |
Registered User
Join Date: Oct 2006
Location: Munich, Germany
Posts: 169
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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:
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 ![]() |
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#14832 | Link |
Moderator
![]() Join Date: Feb 2005
Location: Spain
Posts: 5,660
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@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.
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BeHappy, AviSynth audio transcoder, in Doom9 forums. NicAudio, BassAudio, audio decoders. |
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#14833 | Link | |
Registered User
Join Date: Oct 2006
Location: Munich, Germany
Posts: 169
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Quote:
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. |
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#14834 | Link |
Registered User
Join Date: Oct 2006
Location: Munich, Germany
Posts: 169
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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. Code:
Reducing depth from 64 to 24 bits... |
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#14836 | Link |
Registered User
Join Date: Oct 2006
Location: Munich, Germany
Posts: 169
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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. |
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#14837 | Link |
Registered User
Join Date: Dec 2002
Posts: 5,494
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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.
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#14838 | Link |
結城有紀
Join Date: Dec 2003
Location: NJ; OR; Shanghai
Posts: 620
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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. |
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#14839 | Link |
Registered User
Join Date: Dec 2002
Posts: 5,494
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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" Code:
ffmpeg -i "x.mkv" -map 0:a:0 -af aresample=first_pts=0 "x.wav" |
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