eac3to - audio conversion tool

[Thunderbolt8]

thanks. but its a 7.1 track (BL, BR, C, L, R, SL, SR, LFE), which is the standard order for that then?

[b66pak]

left.wav right.wav center.wav lfe.wav ls.wav rs.wav aux1.wav aux2.wav....it meens l, r, c, lfe, sl, sr, bl, br
_

[bmnot]

Is there/will there be a way to remove pulldown from already made remuxes? I just checked a bunch old HD DVD remuxes in tsmuxer and it detects the video as 29.97fps with pulldown. I tried to output as a Blu-ray disc, but it failed to work. It couldn't remove the pulldown or fix the framerate. Using tsmuxer on a recently remuxed HD DVD, it detects the video as 23.976 and outputs perfect Blu-ray disc structure, perfect for burning. So anyway to fix those old HD DVD remuxes? Unfortunately I sold most of them...

[crazydane]

I have been ripping with eac3to for some time and I have never really had any major issues, but now all of a sudden when I rip, I get the following error regardless of what title I try to rip:

"One of the FLAC encoder's callbacks returned a fatal error"

I was on version 2.78 and just upgraded to 2.80, but the issue remains. I rebooted the machine as well to no avail.

Other than installing the latest Intel graphics/HDMI driver (I have a G45 based motherboard), I have not changed anything recently in my Vista 32-bit environment.

Here are some examples of the error:

The Untouchables:

http://www.cstone.net/~dk/flacenc1.JPG


The Hunt for Red October:

http://www.cstone.net/~dk/flacenc2.JPG

Eac3to errors out after a few minutes of processing, something like 4 "-"s or so, so it's not instant.

I can rip these titles just fine to hd using AnyDVD HD, and then run eac3to on the files on the hd, but why would I be getting this error all of a sudden when pointing eac3to directly to the BD disc in the drive?

Any ideas?

Thanks!

[tebasuna51]

left.wav right.wav center.wav lfe.wav ls.wav rs.wav aux1.wav aux2.wav....it meens l, r, c, lfe, sl, sr, bl, br
_

No! Mus be:
l, r, c, lfe, BackL, BackR, SideL, SideR

sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav SL.wav SR.wav multichannel.wav

[jimz06]

As Beastie Boy suggested, try "eac3to HdDvdSourceFolder movie.mkv". Then play the MKV file just as it is (without the audio). Are there still break ups? If so, probably your playback system is borked (as suggested by Jeff Flowerday). If there are no breaks up when playing the video only MKV, mux the audio track to the video MKV by using mkvtoolnix. Does the final result play fine? If not, again probably something is wrong with your DirectShow filter setup. Try different video/audio decoders and maybe also different video/audio renderers...

Thanks to everyone for the helpful advice. I was trying to create this file to play on a PopcornHour. The one thing I hadn't tried was playing it on the PCH. As it turns out it plays just fine on the PCH:D, so as was indicated it must be the setup on my MPC-HC player.

However, some enlightenment would be appreciated about how this all works. When I used EVOdemux to create the H264 stream I got this:

Complete name :E:\FEATURE_1_MERGED.H.264.stream.0.mpv
Format : AVC
Format/Info : Advanced Video Codec
File size : 22.3 GiB

Video
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings, CABAC : Yes
Format settings, ReFrames : 2 frames
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16/9
Frame rate : 29.970 fps
Standard : Component
Resolution : 24 bits
Colorimetry : 4:2:0
Scan type : Progressive

This played fine on MPC but the fps counter showed 23.97.:confused:

When I muxed this into an mkv (no audio) I got this:

Complete name : E:\ToNMT\Babel.mkv
Format : Matroska
File size : 22.3 GiB
Duration : 2h 23mn
Overall bit rate : 22.2 Mbps
Encoded date : UTC 2008-12-09 00:33:12
Writing application : eac3to
Writing library : Haali DirectShow Matroska Muxer 1.8.122.18

Video
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings, CABAC : Yes
Format settings, ReFrames : 2 frames
Muxing mode : Container profile=Unknown@0.0
Codec ID : V_MPEG4/ISO/AVC
Duration : 2h 23mn
Bit rate : 21.4 Mbps
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16/9
Frame rate : 23.976 fps
Standard : Component
Resolution : 24 bits
Colorimetry : 4:2:0
Scan type : Progressive

Here is the eac3to log:

eac3to v2.80
command line: "E:\eac3to275\eac3to.exe" "D:\Babel\BABEL\HVDVD_TS\FEATURE_1.EVO"+"D:\Babel\BABEL\HVDVD_TS\FEATURE_2.EVO" 3: "E:\ToNMT\Babel.mkv"
------------------------------------------------------------------------------
EVO, 1 video track, 2 audio tracks, 6 subtitle tracks, 2:23:29
"Main Movie"
1: Joined EVO file
2: Chapters, 24 chapters with names
3: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, 1918ms
5: E-AC3, French, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, 1918ms
6: Subtitle, English
7: Subtitle, English, "SDH"
8: Subtitle, French
9: Subtitle, Spanish
10: Subtitle, English, "Forced"
11: Subtitle, French, "Forced"
[v03] Extracting video track number 3...
[v03] Removing h264 pulldown...
[v03] Muxing video to Matroska...
Added fps value to MKV header.
Video track 3 contains 206438 frames.
eac3to processing took 19 minutes, 56 seconds.
Done.

Played back in MPC, I get the regular video break ups using either Haali or the internal splitter.:confused: No problems with other H264 Blu Ray rips. I assume something is going on with that 30 fps to 24 fps that is wreaking havoc on my MPC setup?

[73ChargerFan]

Is there/will there be a way to remove pulldown from already made remuxes? I just checked a bunch old HD DVD remuxes in tsmuxer and it detects the video as 29.97fps with pulldown. I tried to output as a Blu-ray disc, but it failed to work. It couldn't remove the pulldown or fix the framerate. Using tsmuxer on a recently remuxed HD DVD, it detects the video as 23.976 and outputs perfect Blu-ray disc structure, perfect for burning. So anyway to fix those old HD DVD remuxes? Unfortunately I sold most of them...
Try muxing as an m2ts (if not already in that format) and then demux with eac3to. It should remove the pulldown automatically.

[Thunderbolt8]

No! Mus be:
l, r, c, lfe, BackL, BackR, SideL, SideR

sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav SL.wav SR.wav multichannel.wav
aerf, which now, backs first or sides first? the whole process takes quite long :S

[madshi]

Is there/will there be a way to remove pulldown from already made remuxes? I just checked a bunch old HD DVD remuxes in tsmuxer and it detects the video as 29.97fps with pulldown.
You can either try to remux the MKV to m2ts or ts with tsMuxeR (as 73ChargerFan suggested) and then ask eac3to to demux the video or to remux it to MKV. Or if tsMuxeR can't handle the MKV properly, you can demux the video track from the original MKV file by using mkvextract and then you can do "eac3to source.vc1 dest.vc1" to remove the pulldown completely. Please note that mkvextract may need different parameters depending on the video codec. Do a search for mkvextract in this thread to find out which parameters must be used for which video codec. Also for h264 you may need to run the stream through h264info to add AUDs (access unit delimiters) into the h264 stream, or else eac3to won't be able to handle the stream.

I have been ripping with eac3to for some time and I have never really had any major issues, but now all of a sudden when I rip, I get the following error regardless of what title I try to rip:

"One of the FLAC encoder's callbacks returned a fatal error"

I was on version 2.78 and just upgraded to 2.80, but the issue remains. I rebooted the machine as well to no avail.

Other than installing the latest Intel graphics/HDMI driver (I have a G45 based motherboard), I have not changed anything recently in my Vista 32-bit environment.

Here are some examples of the error:

Eac3to errors out after a few minutes of processing, something like 4 "-"s or so, so it's not instant.

I can rip these titles just fine to hd using AnyDVD HD, and then run eac3to on the files on the hd, but why would I be getting this error all of a sudden when pointing eac3to directly to the BD disc in the drive?
I've no idea why it's going wrong. But the fact that it works well when ripping to harddisk first means that it can't be eac3to's fault. Don't know what else to say...

However, some enlightenment would be appreciated about how this all works. When I used EVOdemux to create the H264 stream I got this:
Frame rate : 29.970 fps

When I muxed this into an mkv (no audio)

I got this:
Frame rate : 23.976 fps

Played back in MPC, I get the regular video break ups using either Haali or the internal splitter.:confused: No problems with other H264 Blu Ray rips. I assume something is going on with that 30 fps to 24 fps that is wreaking havoc on my MPC setup?
EvoDemux does not remove the h264 pulldown, so the video is 1080i60. eac3to removes the h264 pulldown, so the video is 1080p24. Don't know why your playback system has problems with that. It seems to work well for everyone else. If all else fails you can tell eac3to to not remove the pulldown by using the "-keepPulldown" switch.

aerf, which now, backs first or sides first? the whole process takes quite long :S
tebasuna51 is usually right. Just as in this case.

[yesgrey]

Which compressing level is used in the conversion to FLAC, the default? Could this value be specified by the user?

Thanks.

[madshi]

Which compressing level is used in the conversion to FLAC, the default? Could this value be specified by the user?
FLAC encoding is hardcoded to max compression level and I don't plan on changing that because I think the compression speed is good enough. So IMHO there's no need to make things more complicated by adding further options just to save a tiny bit of time.

[odin24]

As to why when a DTS track is encoded using Surcode it is 1536kb/s, then when the Zero Padding is removed it is reported as 1510kb/s? MediaInfo and tsMuxeR both report as 1536kb/s.

Another question... whenever I convert a THD track to PCM it always starts as ....24bit.pcm until the actual bit depth is revealed later in the conversion where it reports a constant bit depth of 16 bits. I'm assuming the PCM output will actually be 16 bits... correct?

Thanks for the time.

eac3to v2.78
command line: e3\eac3to f: 1) 2: c:\videos\tdk\tdk.vc1 4: c:\videos\tdk\tdk.thd+ac3 4: c:\videos\tdk\tdk.pcm 4: c:\videos\tdk\tdk.dts 8: c:\videos\tdk\subs1.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:32:13
1: Chapters, 40 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
5: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
6: AC3, French, 5.1 channels, 640kbps, 48khz
7: AC3, Spanish, 5.1 channels, 640kbps, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[s08] Extracting subtitle track number 8...
[a04] Extracting TrueHD stream...
[a04] Extracting TrueHD stream...
[a04] Decoding with libav/ffmpeg...
[a04] Decoding with libav/ffmpeg...
[a04] Swapping endian...
[a04] Writing WAVs...
[a04] Remapping channels...
[a04] Creating file "c:\videos\tdk\tdk.24bit.pcm"...
[v02] Creating file "c:\videos\tdk\tdk.vc1"...
[a04] Creating file "c:\videos\tdk\tdk.thd+ac3"...
[a04] Creating file "c:\videos\tdk\tdk.L.wav"...
[a04] Creating file "c:\videos\tdk\tdk.R.wav"...
[a04] Creating file "c:\videos\tdk\tdk.C.wav"...
[a04] Creating file "c:\videos\tdk\tdk.SR.wav"...
[a04] Creating file "c:\videos\tdk\tdk.LFE.wav"...
[a04] Creating file "c:\videos\tdk\tdk.SL.wav"...
[s08] Creating file "c:\videos\tdk\subs1.sup"...
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
[a04] The original audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...Found Surcode DTS Encoder version 1.0.21.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Video track 2 contains 218979 frames.
eac3to processing took 49 minutes, 32 seconds.
Surcode encoding took 27 minutes, 18 seconds.
Done.

eac3to v2.78
command line: e3\eac3to c:\videos\tdk\tdk.dts c:\videos\tdk\tdk_fixed.dts
------------------------------------------------------------------------------
DTS, 5.1 channels, 2:32:13, 24 bits, 1510kbps, 48khz
Removing DTS zero padding...
Creating file "c:\videos\tdk\tdk_fixed.dts"...
eac3to processing took 2 minutes, 52 seconds.
Done.

[yesgrey]

FLAC encoding is hardcoded to max compression level...

If it's set to the max level no need to change...;) For me the important is not the speed, but the size.
:thanks:

[asarian]

Got a question.

For streaming, the PS3 sadly supports only DD. 2.0 for VC-1 in a WMV container. So, I go to thinking the following: what if you were to 'matrix-encode' five disccrete channels of audio, ala the Dolby ProLogic II scheme, onto a 640kbps DD 2.0 stream in such a way that a Dolby ProLogic II decoder could split those matrixed channels back to the original five channels, would that work? I mean, to take an existing DD 5.1 stream, and turn it into a 2.0 stream that Dolby ProLogic II could then extract more-or-less the same original channels from. Would be a nice trick to get near DD 5.1 sound (whilst it really is just DD 2.0 surround).

Question, of course, is: can eac3to do it? Can it convert a DD 5.1 track to a Dolby ProLogic II 'splittable' 2.0 track? I mean, the -down convert would have to follow the exact Dolby ProLogic II scheme, as if it were a Dolby ProLogic II encoder, as it were.

It's kinda early here (near 6 AM, and I've yet to turn in, LOL), so I may be missing somehing; but with a 640kbps 2.0 stream, I don't see how you couldn't at least get a half-way decent Dolby ProLogic II 5.1 result from it (and thus be able to bypass the need to reencode VC-1 content when streaming).

Feel free to enlighten me (or desillusion me, whichever the case may be).

[madshi]

As to why when a DTS track is encoded using Surcode it is 1536kb/s, then when the Zero Padding is removed it is reported as 1510kb/s? MediaInfo and tsMuxeR both report as 1536kb/s.
The nominal bitrate is 1536kbps. However, the actual bitrate is lower. eac3to calculates & reports the actual bitrate, while MediaInfo and tsMuxeR report the nominal bitrate...

Another question... whenever I convert a THD track to PCM it always starts as ....24bit.pcm until the actual bit depth is revealed later in the conversion where it reports a constant bit depth of 16 bits. I'm assuming the PCM output will actually be 16 bits... correct?
A TrueHD track can have 16bit data or 24bit data or anything in between. eac3to only knows for sure when the whole track is fully decoded. The PCM output consequently is either 16bit (for TrueHD tracks which only contain 16bit worth of audio data) or 24bit (for TrueHD tracks which have 17-24bit of information). eac3to will tell you the bitdepth of the final PCM file.

For me the important is not the speed, but the size.
Agreed.

what if you were to 'matrix-encode' five disccrete channels of audio, ala the Dolby ProLogic II scheme, onto a 640kbps DD 2.0 stream in such a way that a Dolby ProLogic II decoder could split those matrixed channels back to the original five channels, would that work?
Yes. But the end result will not be perfect. It should be quite ok, but keeping the channels separated is of course noticeably better. One problem with DPLII is that the LFE channel is more or less lost.

Question, of course, is: can eac3to do it? Can it convert a DD 5.1 track to a Dolby ProLogic II 'splittable' 2.0 track?
Yes. "-down2".

but with a 640kbps 2.0 stream, I don't see how you couldn't at least get a half-way decent Dolby ProLogic II 5.1 result from it (and thus be able to bypass the need to reencode VC-1 content when streaming).
Half-way decent sounds about right.

[komisar]

madshi
Can you add to eac3to ability for "custom-comman-line-parameters"?
E.g. for neroaacenc:
eac3to.exe input.ac3 output.aac -quality=0.23 -lc

[tebasuna51]

madshi
Can you add to eac3to ability for "custom-comman-line-parameters"?
E.g. for neroaacenc:
eac3to.exe input.ac3 output.aac -quality=0.23 -lc
Encoders can have many parameters.
You can always acces to all command line parameters for a encoder using the pipe option:

eac3to.exe <input> stdout.wav <eac3to_parameters> | <encoder> <full_encoder_parameters> <-> <output>

Example:
eac3to.exe input.ac3 stdout.wav | neroaacenc -ignorelength -q 0.23 -lc -if - -of output.m4a

[komisar]

tebasuna51, Ok. Thnx.

[Thunderbolt8]

got a problem with a TrueHD track here:

50mb sample: http://www.sendspace.com/file/y2diub

eac3to v2.80
command line: eac3to X:\movie 1) 2: G:\movie.mkv 3: G:\movie.flac 3: G:\movie.thd 5: G:\moviecomment1.ac3 6: G:\moviecomment2.ac3 9: G:\movie.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a06] Extracting audio track number 6...
[a05] Extracting audio track number 5...
[s09] Extracting subtitle track number 9...
[a03] This track is not clean.
[a03] This track is not clean.
[a03] Extracting TrueHD stream...
[v02] Muxing video to Matroska...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[libav] Stream parameters not seen; skipping frame
[a03] The libav decoder output an unexpected bitdepth (1).
Aborted at file position 49152.

and when I do it with -nero (nero also decodes TrueHD 5.1 completely lossless, right?) then I get:

G:\eac3to>eac3to X:\movie 1) 2: G:\movie.mkv 3: G:\movie.flac -nero 3: G:\movie.thd 5: G:\moviecomment1.ac3 6: G:\moviecomment2.ac3 9: G:\movie.sup
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a06] Extracting audio track number 6...
[a05] Extracting audio track number 5...
[s09] Extracting subtitle track number 9...
[a03] This track is not clean.
[a03] This track is not clean.
[v02] Muxing video to Matroska...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] Removing TrueHD dialog normalization...
[a05] Removing AC3 dialog normalization...
[a06] Removing AC3 dialog normalization...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Applying RAW/PCM delay...
[a03] This audio track contains more than 16 bits of information.
[a03] Encoding FLAC with libFlac...
[a03] Creating file "G:\movie.thd"...
[a03] Creating file "G:\movie.24bit.flac"...
[a06] Creating file "G:\moviecomment2.ac3"...
[a05] Creating file "G:\moviecomment1.ac3"...
[s09] Creating file "G:\movie.sup"...
[a03] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Original audio track, BL+BR: constant bit depth of 24 bits.
[a03] Processed audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Processed audio track, SL+SR: constant bit depth of 24 bits.
[a03] Audio has a gap of 8ms at playtime 0:00:00.
[a03] The audio file was demuxed without making use of the gap/overlap information.
[a03] Please rerun the same eac3to command line. That will correct the gaps/overlaps.
[a03] Audio has a gap of 8ms at playtime 0:00:00.
[a03] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a03] In order to remove them you'll have to transcode the audio to another format.
Added fps value to MKV header.
Video track 2 contains 178415 frames.
eac3to processing took 59 minutes, 53 seconds.
Done.

and after re-running the cmd line:

eac3to v2.80
command line: eac3to X:\movie 1) 2: G:\movie.mkv 3: G:\movie.flac -nero 3: G:\movie.thd 5: G:\moviecomment1.ac3 6: G:\moviecomment2.ac3 9: G:\movie.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
Audio gap description file detected, will be used for processing...
Audio gap description file detected, can't be used for TrueHD/MLP, though.
[a06] Extracting audio track number 6...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting audio track number 3...
[s09] Extracting subtitle track number 9...
[a05] Extracting audio track number 5...
[a03] This track is not clean.
[a03] This track is not clean.
[v02] Muxing to Matroska...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] Removing TrueHD dialog normalization...
[a03] Removing TrueHD dialog normalization...
[a05] Removing AC3 dialog normalization...
[a06] Removing AC3 dialog normalization...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Applying RAW/PCM delay...
[a03] This audio track contains more than 16 bits of information.
[a03] Encoding FLAC with libFlac...
[a03] Creating file "G:\movie.thd"...
[a03] Creating file "G:\movie.24bit.flac"...
[a05] Creating file "G:\moviecomment1.ac3"...
[a06] Creating file "G:\moviecomment2.ac3"...
[s09] Creating file "G:\movie.sup"...
[a03] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Original audio track, BL+BR: constant bit depth of 24 bits.
[a03] Processed audio track, L+R+C+SL+SR: max 24 bits, average 19 bits.
[a03] Processed audio track, LFE: max 23 bits, average 16 bits.
[a03] Audio has a gap of 8ms at playtime 0:00:00.
[a03] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a03] In order to remove them you'll have to transcode the audio to another format.
Added fps value to MKV header.
Video track 2 contains 178415 frames.
eac3to processing took 1 hour, 1 minute.
Done.

so regarding the difference between the original and the processed track with nero at the re-run, is the resulting flac not lossess or whats going on there?
and when gaps are detected, is there a way to find out if the delay needed is positive or negative? then it would be possible to add that delay to the demuxed truehd track at a later point.

[73ChargerFan]

Report on Mask BD, Extra stream 00024.m2ts using eac3to v2.78
This is the first time I've encountered this, so I thought I'd post it here.

eac3to v2.78
command line: eac3to 2) 2: extra.mkv
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 2 subtitle tracks, 0:27:17
1: Chapters, 2 chapters
2: VC-1, 480p24 /1.001 (3:2) with pulldown flags
3: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
4: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), German
[v02] Extracting video track number 2...
[v02] Removing VC-1 pulldown...
[v02] Muxing video to Matroska...
[v02] Video overlaps for 1 frames at playtime 0:00:00.
[v02] Video overlaps for 1 frames at playtime 0:00:00.
.
.
[v02] Video overlaps for 1 frames at playtime 0:34:01.
[v02] The MKV file was created without making use of the gap/overlap information.
[v02] Please check whether audio is in sync. If it is in sync everything is fine.
[v02] Otherwise you can ask eac3to to repeat the muxing. It will then automatically
[v02] make use of the detailed gap/overlap information.
Added fps value to MKV header.
Video track 2 contains 49058 frames.

gave 9684 "video overlaps" messages, or 1 every 10 seconds,
and a playtime of 0:34:06, which is precisely 30/24 times the correct playtime 0:27:17.

Running the command second time used the gaps file and the new mkv file has the correct video length. :D

The playtime of the first mkv file was checked using mpc-hc 918 and MediaInfo 0.7.7.4.

---------------------------------------------

This number of video overlaps in a file necessitates a second run.

fyi

---------------------------------------------
edit:

This didn't work as I thought. The video jitters and jutters every few seconds.

[digitlman]

Thanks for the great tools. they work just wonderfully! however i have a dilemma. I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio. i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?

thanks

[deathlord]

madshi,
Got a problem with The Simpsons Movie BD (eac3to 2.80.0.0):
eac3to 1) 2: g:\Simpsons\Video.mkv 3: g:\Simpsons\Engli
sh_5.1_dts-hdma.flac -down16
M2TS, 1 video track, 9 audio tracks, 24 subtitle tracks, 1:41:28
1: Chapters, 25 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: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
5: DTS, German, 5.1 channels, 24 bits, 768kbps, 48khz
6: AC3, Dutch, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, Dutch, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: AC3, Finnish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
9: AC3, Swedish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
10: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
11: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
12: Subtitle (PGS), English
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), French
16: Subtitle (PGS), German
17: Subtitle (PGS), German
18: Subtitle (PGS), Dutch
19: Subtitle (PGS), Dutch
20: Subtitle (PGS), Finnish
21: Subtitle (PGS), Finnish
22: Subtitle (PGS), Swedish
23: Subtitle (PGS), Swedish
24: Subtitle (PGS), English
25: Subtitle (PGS), French
26: Subtitle (PGS), German
27: Subtitle (PGS), Dutch
28: Subtitle (PGS), Finnish
29: Subtitle (PGS), Swedish
30: Subtitle (PGS), English
31: Subtitle (PGS), French
32: Subtitle (PGS), German
33: Subtitle (PGS), Dutch
34: Subtitle (PGS), Finnish
35: Subtitle (PGS), Swedish
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "g:\Simpsons\English_5.1_dts-hdma.flac"...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] The h264 muxer doesn't support this stream type yet.
[v02] Please send a 20MB sample to dear@madshi.net
Aborted at file position 22046294016.

Do you need a sample, and if yes, what exactly should I make the sample from?

Also, this movie is supposed to have force subs. But how do I see which it is?

[madshi]

got a problem with a TrueHD track here:

50mb sample: http://www.sendspace.com/file/y2diub

[a03] This track is not clean.
[a03] Decoding with libav/ffmpeg...
[libav] Stream parameters not seen; skipping frame
[a03] The libav decoder output an unexpected bitdepth (1).
Aborted at file position 49152.
Obviously the source file is not clean. The libav decoder doesn't seem to be able to handle this situation. Are you sure that the rip is not corrupt? I'd strongly suggest reripping the disk.

and when I do it with -nero (nero also decodes TrueHD 5.1 completely lossless, right?) then I get:

and after re-running the cmd line:

Audio gap description file detected, will be used for processing...
[a03] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Original audio track, BL+BR: constant bit depth of 24 bits.
[a03] Processed audio track, L+R+C+SL+SR: max 24 bits, average 19 bits.
[a03] Processed audio track, LFE: max 23 bits, average 16 bits.

so regarding the difference between the original and the processed track with nero at the re-run, is the resulting flac not lossess or whats going on there?
I've recently added a post processing filter which smooths the audio cut points where gaps/overlaps are fixed to remove audio spikes. This post processing filter creates 24bit samples. That's why the bitdepth analyzation reports different results after the gaps have been fixed. In your specific case that's no problem cause the stream was overall (up to) 24bit, anyway. So it's still practically lossless. But for 16bit the behavior of the smoothing filter is not very good. So I'll change it for the next build so that it only creates 16bit samples.

and when gaps are detected, is there a way to find out if the delay needed is positive or negative? then it would be possible to add that delay to the demuxed truehd track at a later point.
When there's a gap you need to apply a positive delay. For overlaps you need to apply a negative delay. The gap/overlap amount (in milliseconds) is listed in the eac3to log of the first run.

Report on Mask BD, Extra stream 00024.m2ts using eac3to v2.78
This is the first time I've encountered this, so I thought I'd post it here.

2: VC-1, 480p24 /1.001 (3:2) with pulldown flags
gave 9684 "video overlaps" messages, or 1 every 10 seconds,
and a playtime of 0:34:06, which is precisely 30/24 times the correct playtime 0:27:17.
I'm aware of this problem. It occurs with VC-1 extras which are encoded as 480p30 with pulldown flags to 480i60. eac3to incorrectly identifies these as "480p24 with pulldown flags". So basically a wrong framerate is detected which results in the problems you noticed. I have on my to do list to fix this. It only occurs with some VC-1 extras, though, so it's not too bad...

I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio.
So please ask Syabas and Popcorn Hour to add support for MKV files with multichannel FLAC in it. It's about time they finally support that!

i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?
You can use mkvextract the demux the video and audio tracks. Then you can use eac3to to convert the FLAC to whatever format you need. Can't help you with m2ts muxing. That's outside the scope of this thread.

Got a problem with The Simpsons Movie BD (eac3to 2.80.0.0):
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] The h264 muxer doesn't support this stream type yet.
[v02] Please send a 20MB sample to dear@madshi.net
Aborted at file position 22046294016.
That's weird. Especially the PTS break before the muxing failure report. My best guess is that your source is corrupt and that the corruption is responsible for the problems. Can you please rerip the disk? Please rip to harddisk first and then run eac3to on the folder on harddisk. This is known to be the most stable way if you have problems with corruption.

Has anyone else remuxed the Simpsons movie with eac3to yet? Did it work ok for you?

[Jeff Flowerday]

Thanks for the great tools. they work just wonderfully! however i have a dilemma. I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio. i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?

thanks

If your flac was created from the lossless soundtrack at this point your only choice is to convert the audio to LPCM before stuffing it in a m2ts container. Unfortunaty your file size will increase dramatically, I've seen 4-7GB per file.

I ended up dumping the popcorn hour and putting a HTPC in it's place. With madshi's madflac directshow filter, my MKV's work flawlessly and the Media Portal interface is sooooooo much richer than the popcorn hour and/or the YAMJ plugin for iHome.

[rack04]

Madshi,

Could you lend any advice to neuron2 regarding how to detect dts-hd ma, dts-hd hr, and truehd?

http://forum.doom9.org/showthread.php?p=1223545#post1223545

Thanks.

[rica]

Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.

[nautilus7]

I don't think it can be done with eac3to. Use a hex editor instead.

[rica]

Oooo, i hate hex editors.
Thanks anyways.

[banker_rishad]

Madshi sir i tried converting vob to dts but the output is dts that is running away. I mean the audio is running or the tempo is high. what to do. plz advise.

[yonta]

Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.

tail.exe (http://www.mediafire.com/?sharekey=734a338392b0f703d2db6fb9a8902bda) can do that.

[Snowknight26]

Or clip.exe (http://www.stfcc.org/misc/clip.exe).

[asarian]

Is your issue. TsMuxer don't accept lpcm files. These files are raw audio data without header and can't be recognized out of a container than inform about bitdepth, channels, samplerate and endian.

Select wav like output file and can be recognized by TsMuxer if is <4GB (probably because 2 C and 16 bit). For wav files > 4GB (5.1 and > 130 min.) you need pcm output and Pcm2Tsmu.

This LPCM conversion is very exciting. :) I'm still not getting it entirely right. Here's an example:


eac3to c:\video\20000.m2ts 4: c:\video\wall-e.pcm
M2TS, 3 video tracks, 3 audio tracks, 3 subtitle tracks, 1:37:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: AC3 Surround, 2.0 channels, 192kbps, 48khz
6: AC3 Surround, 2.0 channels, 192kbps, 48khz
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "c:\video\wall-e.pcm"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 140154 frames.
Video track 2 contains 140154 frames.
Video track 3 contains 140154 frames.
eac3to processing took 13 minutes, 20 seconds.
Done.


So far so good (?). Now:


Pcm2Tsmu c:\video\wall-e.pcm c:\video\wall-e-lpcm.pcm -c 6


This kinda works. I get an audible 5.1 track (of around 5G), but a lot seems missing: like if the front channels were gone. Mind you, there's sound in the front channels, but it's almost as if things are miswired, as you can hear voices only very faintly, and other major stuff. Weird.

Maybe this has to do with the -16 switches (both for eac3to and Pcm2Tsmu). But if I use those, like:


eac3to c:\video\20000.m2ts 4: c:\video\wall-e.pcm -16
Pcm2Tsmu c:\video\wall-e.pcm c:\video\wall-e-lpcm.pcm -i 16 -c 6


Then my PS3 just hangs at the start of the movie (doesn't even start playing the video). So, my question is, how is this down-converting to 16bit supposed to go then? Or, in case down-converting isn't necessary, why am I missing so much audio on the 24bit mix? Other than clearly some major channels/sound missing, what is being output actually sound superb, btw. So I guess I'm close. :)

Thanks

[asarian]

^^ Seems I'm in bad luck. I send the LPCM over a Toslink, and apparently that only does stereo (I thought it would do 5.1 LPCM as well). Alas.

Interestingly, though, the PS3 actually lists the audio as "Linear PCM, 5.1ch, 6.9Mbps", while streaming (via Twonky). The latter is remarkable, as I always thought, until now, that 640kbps AC3 was the limit for streaming on the PS3. And now it turns out it does full 5.1 LPCM, and at near 7Mbps even! (converted from the DTS-HD track).

@Madshi: still, I'm not a man without hope, as I can now still try my Dolby ProLogic II trick; this time, however, not with a measly 640kbps AC3 stream, but with a 7Mbps LPCM stereo stream: 11x as much bandwidth. :)

Still, seems the PS3 streams a whole lot more than 640kbps AC3, after all. And that is good news. That I just need to buy a better amp is irrelevant here. :)

[n0mag!c]

Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.
To cut last part of any file you can use CutTools (http://www.softpedia.com/get/System/File-Management/CutTools.shtml).
It works in primitive binary mode and doesn't know anything about frames or packets of specific file types. You can calculate cut point for yourself if you know frame or packet length.

[telmoMRC]

some days ago i have convert one dts audio to ac3 384 but now doesn´t work the dts audio [input.dts] is in the same directory the eac3to and give this error source file "input.dts" not found wy do not work ???

screen
http://e.imagehost.org/0702/nao_d.jpg

[madshi]

i tried converting vob to dts but the output is dts that is running away. I mean the audio is running or the tempo is high.
I don't really know what you mean. And posting attachments to this forum doesn't make sense, unfortunately, because it usually takes years until they are approved.

This LPCM conversion is very exciting. :) I'm still not getting it entirely right.
Both Pcm2Tsmu and tsMuxeR have their own threads. Please ask there for help. I cannot help you and this topic doesn't really belong here.

some days ago i have convert one dts audio to ac3 384 but now doesn´t work the dts audio [input.dts] is in the same directory the eac3to and give this error source file "input.dts" not found wy do not work ???
It seems that your OS is configured in such a way that the file extensions of files which are known to the OS are not displayed. So my best guess is that the file's real name is "input.dts.dts".

[madshi]

Can you add to eac3to ability for "custom-comman-line-parameters"?
E.g. for neroaacenc:
eac3to.exe input.ac3 output.aac -quality=0.23 -lc
Not sure how many funny parameters neroaacenc has. Right now I'd have to implement every single one by hand to make them all work. That's no fun, so I'm probably not gonna do it. I might sooner or later find a way to automatically pass all unknown parameters to the encoder module. Then what you're looking for would automatically work.

[telmoMRC]

It seems that your OS is configured in such a way that the file extensions of files which are known to the OS are not displayed. So my best guess is that the file's real name is "input.dts.dts".


i try that and doesn´t work what can i do ?

[rica]

tail.exe (http://www.mediafire.com/?sharekey=734a338392b0f703d2db6fb9a8902bda) can do that.

Or clip.exe (http://www.stfcc.org/misc/clip.exe).

Thanks guys.

Can you please give me a command line sample for tail.exe and clip.exe?

usage: tail <infile> <outfile> <length>


For tail i gave it a go with:

C:\>clip\tail E:\BDMV\stream\00181.m2ts C:\HD\out.m2ts 500MB


But nothing has happened.

For clip.exe...???

usage: clip <infile> <outfile> <start offset> <length>

***madshi, thanks for your understanding.

[asarian]

Both Pcm2Tsmu and tsMuxeR have their own threads. Please ask there for help. I cannot help you and this topic doesn't really belong here.


Fortunately, for me, I have neither a tsMuxeR, nor a Pcm2Tsmu issue. :) I just like to know one eac3to thing; when I do this:

eac3to 20000.m2ts 4: wall-e.pcm -down2

eac3to lowers the bitrate to about 2Mbps. Any way I can keep the original bitrate for the two channels?

[yonta]

If you want to cut out the last 50MBytes

tail.exe infile outfile 50000000

[digitlman]

i dunno which place to post this as i am not sure which program is the problem but i will start here...

i have tried this with 2 different mkv+flac files and neither flac file is recognized by eac3to. is there an option with eac3to i need to change or is it something wrong with my mkvextract options?

mediainfo recognizes the flac files just fine, says they are 6channel 24bit 48000hz. i also tried using the --no-ogg option with same results

i am just trying to convert the flac file to lpcm.


C:\Program Files\MKVtoolnix>mkvextract tracks "f:\Pirates Of The Caribbean 1 AVC
LPCM24.mkv" 1:d:\p1.h264 2:d:\p1.flac
Extracting track 1 with the CodecID 'V_MPEG4/ISO/AVC' to the file 'd:\p1.h264'.
Container format: AVC/h.264 elementary stream
Extracting track 2 with the CodecID 'A_FLAC' to the file 'd:\p2.flac'. Container
format: Ogg (FLAC in Ogg)
progress: 100%

C:\Program Files\MKVtoolnix>

C:\eac3to>eac3to d:\p1.flac
The format of the source file could not be detected.

C:\eac3to>

C:\eac3to>eac3to d:\cars.flac h:\cars3.pcm
The format of the source file could not be detected.

[rica]

If you want to cut out the last 50MBytes

tail.exe infile outfile 50000000

Thanks, i'd like to have a sample CMD line of clip.exe as well?

"start offset" ?

[yonta]

You can cut out 1920000 bytes from a file starting from the 193rd byte.
clip infile outfile 193 1920000

[madshi]

i try that and doesn´t work what can i do ?
First of all configure your OS correctly. Letting the OS hide known extensions is the best way to confuse yourself.

I just like to know one eac3to thing; when I do this:

eac3to 20000.m2ts 4: wall-e.pcm -down2

eac3to lowers the bitrate to about 2Mbps. Any way I can keep the original bitrate for the two channels?
The bitrate for PCM is a simple calculation:

bitrate = bitDepth * channels * sampleRate

So if you go from 6 to 2 channels, obviously the bitrate decreases. There's nothing you can do about that, and there's no reason to do anything about it. Bitrate doesn't matter with PCM, it only matters for lossy compression.

i dunno which place to post this as i am not sure which program is the problem but i will start here...

i have tried this with 2 different mkv+flac files and neither flac file is recognized by eac3to. is there an option with eac3to i need to change or is it something wrong with my mkvextract options?

mediainfo recognizes the flac files just fine, says they are 6channel 24bit 48000hz. i also tried using the --no-ogg option with same results
You do have to use the "no ogg" option, because eac3to doesn't support the OGG container. My best guess is that you put the option at the wrong place in the mkvextract command line. That has happened before...

Can you please give me a command line sample for tail.exe and clip.exe?
I hope that you are asking these questions because you want to make a sample for me? Because if not you're once again very much out of topic here...

[asarian]

The bitrate for PCM is a simple calculation:

bitrate = bitDepth * channels * sampleRate

So if you go from 6 to 2 channels, obviously the bitrate decreases. There's nothing you can do about that, and there's no reason to do anything about it.

The reason to keep the high bitrate was to use the Dolby Prologic II trick, so as to use the full 6.9Mbps for the 2 channels.

Anyway, looks like I'll have to buy the Marantz SR-8002 soon, and do things properly. :)


Bitrate doesn't matter with PCM, it only matters for lossy compression.

? I thought bitrate always matters. I mean, isn't a 8Mbps LPCM stream better than a 640kpbs one?

[madshi]

? I thought bitrate always matters. I mean, isn't a 8Mbps LPCM stream better than a 640kpbs one?
Have you tried to understand the LPCM bitrate formula I gave you? If so, you would have understood by now that the bitrate you got is lower because you decreased the number of channels. The only way to increase the bitrate again is to either increase the samplerate or the bitdepth. Both of which would not improve audio quality in your case. That's the last thing I'll say on this topic.

[kurt]

how to demux mpeg2 video properly?

eac3to xxx.m2ts video.m2v does not work (audio conversion not supported). neither than mpeg2, mpeg, mpg.....

(I know there is the -demux switch)

[nautilus7]

Well, use the track number format. Something like:

eac3to xxx.m2ts 2: video.m2v

[madshi]

how to demux mpeg2 video properly?

eac3to xxx.m2ts video.m2v does not work (audio conversion not supported). neither than mpeg2, mpeg, mpg.....
It should work the way you did it. Please post the full eac3to log (between [ code ][ / code ]).