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26th January 2015, 05:35 | #12961 | Link | |
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Code:
ffmpeg -i 00000.m2ts -map 0:1 extract.wav eac3to extract.wav extract.dts |
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26th January 2015, 14:02 | #12962 | Link |
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Post about ffmpeg moved to ffmpeg syntax thread.
Please don't use this thread for ffmpeg discussions.
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31st January 2015, 05:12 | #12963 | Link |
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I'm experimenting with the dolby headphone wrapper plugin that's available with foobar, so I'm remapping channels with eac3to to see the effects. My source file is the song from the end of the first hobbit movie (7.1 channel dts-hd ma). I downmix to 5.1 and then analyze the channels with the foobar oscilloscope. The default output of eac3to is L,R,C,LFE,SL,SR, and is verified by the waveforms. The loudest is the center, followed by the left and right, then the two surrounds, and the LFE is near non existent, but is there.
http://imgur.com/DeIVxyd I did the same thing but remapped the channels to L,R,SL,SR,LFE,C. As can be seen in the oscilloscope, the center is still the loudest (6th waveform), followed by the left and right (first and second waveforms), but the LFE somehow has more activity now and is louder (5th waveform). The SL and SR are closely related in activity and amplitude to each other, just like before remapping, but have lower activity and amplitude than the LFE... and also have lower amplitude than before the remap. http://imgur.com/mTc9GgA What's going on? Does eac3to apply different mix levels while disregarding which channel is what and going just based on sequential order, or is foobar doing something. Last edited by Arm3nian; 31st January 2015 at 05:15. Reason: more info :) |
31st January 2015, 11:34 | #12964 | Link | |||
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31st January 2015, 13:26 | #12965 | Link |
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Hi everybody,
I've improved StaxRip's eac3to support quite much lately, maybe a few people here have some interest to look at the latest release: http://forum.doom9.org/showthread.ph...91#post1707791 |
1st February 2015, 00:51 | #12966 | Link | |
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So if the channels are not in the default order when remapped then they get mixed at wrong levels? Last edited by Arm3nian; 1st February 2015 at 01:43. |
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1st February 2015, 19:17 | #12967 | Link |
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eac3to input71.dt output.wav -down6
finish with: Code:
FL' = FL (0) FR' = FR (1) FC' = FC (2) LF' = LF (3) SL' = BL + SL (4 + 6) SR' = BR + SR (5 + 7) finish with: Code:
FL' = FL (0) FR' = FR (1) FC' = BL (4) LF' = BR (5) SL' = LF + SL (3 + 6) SR' = FC + SR (2 + 7) Maybe you need 2 pass, first to only -down6 and after remap with -0,1,4,5,3,2
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6th February 2015, 03:05 | #12968 | Link | |
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11th February 2015, 13:26 | #12969 | Link |
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6.1 flac channel order
This seems to be a semi-popular topic, but I don't think my specifc question has previously been answered.
With lord of the rings 6.1 dts-hd-ma audio Code:
C:\> eac3to lotr.mkv MKV, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:07:41, 24p /1.001 1: h264/AVC, English, 1080p24 /1.001 (16:9) 2: DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz (core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz) "Surround 6.1" ... Code:
C:\> eac3to lotr.mkv 1:lord_of_the_rings.flac I have verified that the flac is bit-for-bit identical (including channel order) to the makemkv decode results. This post on the makemkv forum says that the channel order should therefore be L R C LFE Ls Rs Cs However, if I produce my own 7 channel flac file (with libsoundfile/audiolab), I find that downmixing it with eac3to Code:
eac3to my_7_channel.flac my_7_channel.stereo.flac -downstereo Code:
[[ 1. 0. 0.7071 0. 0.7071 1. 0. ] [ 0. 1. 0.7071 0. 0.7071 0. 1. ]] L R C LFE Cs Ls Rs Is this discrepancy due to differing channel mask between my file and that created by eac3to? Or is my first conclusion incorrect that eac3to places the centre surround channel last in the flac when decoding from dts-hd-ma? |
11th February 2015, 13:50 | #12970 | Link | |
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The "official" FLAC 6.1 order is L R C LFE Cb Ls Rs .. unfortunately 6.1 is annoying and both orders are commonly used.
See here: http://xiph.org/flac/format.html#frame_header (scroll down to <4> channel assignment) Quote:
However, it may be nice and write metadata into the FLAC headers to indicate the channel order, which eac3to should hopefully read.
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11th February 2015, 19:06 | #12971 | Link | |
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I tested MakeMKV decoding to wav a DTS-MA 6.1 and output the standard 6.1 channel order (the same than eac3to output): FL FR FC LF BC SL SR where BC = Back Center SL = Side Left SR = Side Rigth Talking about channel order please use always the channel names and order defined here to avoid confusions. The channel names and order inside a encoded format can be differents but, to transcode, we needs always decode to standard WAV order and channels, to allow the new encoder convert the audio to their internal order and channels. Standard order for 7.1 is (like I put in my precedent post): FL FR FC LF BL BR SL SR
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 11th February 2015 at 19:09. |
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13th February 2015, 17:17 | #12972 | Link |
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regarding 2.1 DTS-HD MA tracks: Does the arcsoft bug (LFE channel cannot be decoded) still exist in each arcsoft version? Or is there a version which is not affected by the bug?
when I play such a track with MPC-HC & LAV Audio with arcsoft capabilities, I wont hear the LFE channel then? what happens when I transform the 2.1 track with eac3to, using the arcsoft decoder, e.g. into wavs. will the LFE channel then be processed incorrectly? but when I just demux the track nothing happens?
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13th February 2015, 18:07 | #12973 | Link |
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Code:
M2TS, 1 video track, 2 audio tracks, 5 subtitle tracks, 24p /1.001 1: Chapters, 24 chapters 2: h264/AVC, 1080p24 /1.001 (16:9) 3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz) 4: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz 5: Subtitle (PGS), English 6: Subtitle (PGS), Swedish 7: Subtitle (PGS), Norwegian 8: Subtitle (PGS), Danish 9: Subtitle (PGS), Finnish a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead. a03 The libav DTS decoder doesn't decode the full DTS-HD information. Creating file "V:\Blurays and DVDs\Saving Mr Banks\Saving Mr Banks_Chapters.txt"... v02 Extracting video track number 2... a03 Extracting audio track number 3... a03 Extracting DTS core... a03 Decoding with libav/ffmpeg... a03 Reducing depth from 64 to 32 bits... a03 Encoding AAC <0.50> with NeroAacEnc... v02 Creating file "V:\Blurays and DVDs\Saving Mr Banks\Saving Mr Banks.h264"... a03 0:21:12 The source file seems to be damaged (sync byte missing). v02 0:21:13 The source file seems to be damaged (sync byte missing). v02 0:21:13 The source file seems to be damaged (discontinuity). a03 0:21:12 The source file seems to be damaged (discontinuity). a03 Currently it's not possible that GUIs can read colored eac3to output, because of this I don't have the full error message. |
14th February 2015, 22:21 | #12976 | Link |
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Does the DRC bug still exist in 3.27?
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14th February 2015, 23:06 | #12977 | Link | |
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Now the default decoder for AC3 is libav instead Nero for that problem.
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14th February 2015, 23:20 | #12978 | Link |
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tebasuna51, using libav leads to another issue, doesn't it?
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15th February 2015, 01:19 | #12979 | Link | |
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15th February 2015, 11:22 | #12980 | Link |
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Another question about the Arcsoft DTS decoder. Is it better to use it when decoding "regular" 5.1 DTS tracks? Or is it useful only for the recent 7.1 evolutions? AFAIK, a decoder that is able to decode a specific format (without bugs) should give exactly the same result than another decoder, and therefore there is no advantage to use the slow Arcsoft decoder when decoding a DTS tracks that the libav decoder can also decode properly. Am I right?
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