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14th April 2017, 22:08 | #14203 | Link | |
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Cmd on the other hand doesn't have a replace-tool, so you'd have to use 3rd party software. My favorite: Code:
eac3to.exe | xidel.exe -s - -e "replace($raw,'[\b]','','!')" | clip or eac3to.exe | xidel.exe -s - -e "replace($raw,'\x08','','!')" | clip Code:
eac3to.exe | sed.exe "s/\x08//g" | clip
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18th April 2017, 12:10 | #14204 | Link |
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Hi, I was under the impression that when converting a 7.1 track to 5.1, eac3to would automatically run a second pass and reduce the volume if clipping was detected. So I just ran this conversion and eac3to didn't generate a second pass. As a test, I ran the same command with -normalize added, and eac3to applied a -0.09db gain, so in other words, there was clipping. So am I supposed to add -normalize to every 7.1 to 5.1 downmix?
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18th April 2017, 12:21 | #14205 | Link |
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I can't tell you for sure, but I guess there may be circumstances where eac3to will only extract a 5.1 core and simply omit additional channels. Or it was just matrix encoded, no discrete 7.1 source anyway. You will have to specify more precisely which source format you have and which parameter set you used.
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18th April 2017, 12:28 | #14206 | Link | |
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18th April 2017, 12:55 | #14208 | Link | |
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So I guess eac3to just applies a standard -0.09 gain when downmixing 7.1 to 5.1 when the -normalize switch is used, and no clipping is detected? So now of course I'm wondering, is using the -normalize switch good practice, or should I just let eac3to mix the surrounds and only automatically normalize if there is detected clipping? |
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18th April 2017, 13:05 | #14209 | Link | ||||
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Even, if the automatic normalize show a big attenuation value, you can consider use a special downmix (MeGUI or BeHappy) than preserve volume in FL,FR,FC and LFE, and only apply attenuation to mix SL+BL, SR+BR.
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18th April 2017, 13:20 | #14210 | Link | |
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18th April 2017, 15:05 | #14211 | Link |
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Is there a way to avoid the re-encoding of the AC3 embedded track when processing a segmented movie?
Code:
[a03]eac3to v3.31 command line: "D:\Archivos de Programa\EAC32AC3\eac3to.exe" "I:\TestTHD\BDMV\PLAYLIST\00620.mpls" 1) 3: "D:\\00620.mpls_3eng.thd+ac3" -progressnumbers -log="D:\Archivos de Programa\EAC32AC3\UsEac3To.log" ------------------------------------------------------------------------------ M2TS, 1 video track, 6 audio tracks, 4 subtitle tracks, 2:07:47, 24p /1.001 1: Chapters, 16 chapters 2: h264/AVC, 1080p24 /1.001 (16:9) 3: TrueHD/AC3 (Atmos), English, 7.1 channels, 48kHz (embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz) 4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, -9ms 5: AC3, French, 5.1 channels, 640kbps, 48kHz, -9ms 6: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms 7: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms 8: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms 9: Subtitle (PGS), English 10: Subtitle (PGS), English 11: Subtitle (PGS), Spanish 12: Subtitle (PGS), French [a03] AC3 encoding doesn't support back channels. Will mix them into the surround. [a03] Extracting audio track number 3... [a03] Extracting audio track number 3... [a03] Extracting TrueHD stream... [a03] Extracting TrueHD stream... [a03] Decoding with libav/ffmpeg... [a03] Mixing surround channels... [a03] Remapping channels... [a03] Encoding AC3 <640kbps> with libAften... [a03] Creating file "D:\\00620.mpls_3eng.thd+ac3"... [a03] Audio overlaps for 5ms at playtime 0:15:21. <WARNING> [a03] Audio overlaps for 5ms at playtime 0:48:26. <WARNING> [a03] The audio gaps/overlaps can't be removed from the TrueHD bitstream. <WARNING> [a03] In order to remove them you'll have to transcode to another format. <WARNING> [a03] Original audio track, L: max 23 bits, average 19 bits. [a03] Original audio track, R+BL+BR: max 24 bits, average 18 bits. [a03] Original audio track, C: max 22 bits, average 19 bits. [a03] Original audio track, LFE: constant bit depth of 18 bits. [a03] Original audio track, SL: max 21 bits, average 18 bits. [a03] Original audio track, SR: max 20 bits, average 18 bits. [a03] Processed audio track, L: max 23 bits, average 19 bits. [a03] Processed audio track, R+SL+SR: max 24 bits, average 19 bits. [a03] Processed audio track, C: max 22 bits, average 19 bits. [a03] Processed audio track, LFE: constant bit depth of 18 bits. Video track 2 contains 183851 frames. eac3to processing took 6 minutes, 58 seconds. Done. |
7th May 2017, 12:14 | #14213 | Link | |
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Quote:
it doesn't crash if there's no WAV file.
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8th May 2017, 15:55 | #14215 | Link |
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That is, I don't believe there is a "command" to disable the notification. It's just designed that way.
Hence: Delete the WAV file in the eac3to folder. No WAV file, no sound. It doesn't break anything if you delete it. Try looking inside the folder with eac3to, or whatever GUI/AiO you are using and delete the WAV files. They're not really needed.
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8th May 2017, 19:23 | #14216 | Link |
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I do that right away, after the first success sound reminds me I rename the wavs to nos. It is actually a nice and easy way to configure that feature.
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12th May 2017, 21:12 | #14217 | Link |
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Looks like eac3to writes chapters to m4a or is it created by NeroAacEnc?
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Audio encoding using eac3to 3.31 x86 -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- D:\Projekte\VS\VB\StaxRip\bin\Apps\eac3to\eac3to.exe "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2 44ms.flac" "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a" -quality=0.35 -normalize +44ms -progressnumbers FLAC, 2.0 channels, 0:00:41, 24 bits, 827kbps, 44.1kHz Decoding FLAC... Applying RAW/PCM delay... Writing WAV... Creating file "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a.pass1.wav"... The original audio track has a constant bit depth of 24 bits. Starting 2nd pass... Reading WAV... Reducing depth from 64 to 32 bits... Encoding AAC <0.35> with NeroAacEnc... Applying 0.05dB gain... The processed audio track has a constant bit depth of 32 bits. eac3to processing took 1 second. Done. Start: 21:53:24 End: 21:53:25 Duration: 00:00:01 General Complete name : D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a Format : MPEG-4 Format profile : Base Media / Version 2 Codec ID : mp42 (M4A /mp42/isom) File size : 318 KiB Duration : 41 s 215 ms Overall bit rate mode : Variable Overall bit rate : 63.2 kb/s Encoded date : UTC 2017-05-12 19:53:25 Tagged date : UTC 2017-05-12 19:53:25 Audio ID : 1 Format : AAC Format/Info : Advanced Audio Codec Format profile : LC Codec ID : 40 Duration : 41 s 215 ms Bit rate mode : Variable Bit rate : 61.4 kb/s Maximum bit rate : 70.0 kb/s Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 44.1 kHz Frame rate : 43.066 FPS (1024 spf) Compression mode : Lossy Stream size : 309 KiB (97%) Writing library : Nero AAC codec 1.5.4.0 Encoding settings : -q 0.35 Encoded date : UTC 2017-05-12 19:53:25 Tagged date : UTC 2017-05-12 19:53:25 Menu -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Muxing using mkvmerge 11.0.0 pre x64 -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- D:\Projekte\VS\VB\StaxRip\bin\Apps\MKVToolNix\mkvmerge.exe -o "D:\Video\Samples\MKV\720p SBR_new.mkv" "D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264" --audio-tracks 0 --language 0:ger --default-track 0:0 "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a" --global-tags "D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_tags.xml" --ui-language en mkvmerge v11.0.0 ('Alive') 64bit 'D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264': Using the demultiplexer for the format 'AVC/h.264'. 'D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a': Using the demultiplexer for the format 'QuickTime/MP4'. 'D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264' track 0: Using the output module for the format 'AVC/h.264 (unframed)'. 'D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a' track 0: Using the output module for the format 'AAC'. The file 'D:\Video\Samples\MKV\720p SBR_new.mkv' has been opened for writing. The cue entries (the index) are being written... Multiplexing took 0 seconds. Start: 21:53:36 End: 21:53:37 Duration: 00:00:00 General Complete name : D:\Video\Samples\MKV\720p SBR_new.mkv Format : Matroska Format version : Version 4 / Version 2 File size : 11.4 MiB Duration : 41 s 156 ms Overall bit rate : 2 324 kb/s Encoded date : UTC 2017-05-12 19:53:36 Writing application : mkvmerge v11.0.0 ('Alive') 64bit Writing library : libebml v1.3.4 + libmatroska v1.4.5 Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : Baseline@L2.1 Format settings, CABAC : No Format settings, ReFrames : 1 frame Codec ID : V_MPEG4/ISO/AVC Duration : 41 s 66 ms Bit rate : 2 264 kb/s Width : 480 pixels Height : 272 pixels Display aspect ratio : 16:9 Frame rate mode : Constant Frame rate : 30.000 FPS Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.578 Stream size : 11.1 MiB (97%) Writing library : x264 core 148 r2762 90a61ec Default : Yes Forced : No Audio ID : 2 Format : AAC Format/Info : Advanced Audio Codec Format profile : LC Codec ID : A_AAC Duration : 41 s 146 ms Bit rate : 61.5 kb/s Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 44.1 kHz Frame rate : 43.066 FPS (1024 spf) Compression mode : Lossy Delay relative to video : 10 ms Stream size : 309 KiB (3%) Language : German Default : No Forced : No Menu 00 : 00:00.059 : en:00:00:00.059
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12th May 2017, 21:54 | #14219 | Link |
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And mkvmerge accounts this delay? It's shown by a player as chapters (Editions):
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12th May 2017, 22:15 | #14220 | Link |
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NeroAacEnc writes the delay info 2 times:
A) as chapter B) as proprietary iTunSMPB tag Mkvmerge reads and applies delay from B automatically. A is treated like a regular chapter. So probably best to set --no-chapters for these m4a files. |
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