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Old 28th September 2014, 18:11   #12821  |  Link
tebasuna51
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Can be used to decode the track but can't be used by eac3to until madshi can parse the track and add new libav dll's.

By the moment you can do, for instance:

ffmpeg -i 000000.m2ts -map 0:1 recoded.flac
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Old 28th September 2014, 20:27   #12822  |  Link
maxmercy
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Quote:
Originally Posted by tebasuna51 View Post
Can be used to decode the track but can't be used by eac3to until madshi can parse the track and add new libav dll's.

By the moment you can do, for instance:

ffmpeg -i 000000.m2ts -map 0:1 recoded.flac
Will this mean a new version of eac3to and the ability to extract the 7.1 track from the DTHD stream? Looking forward to it!

JSS
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Old 29th September 2014, 05:47   #12823  |  Link
JimmyBarnes
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Quote:
Originally Posted by tebasuna51 View Post
Can be used to decode the track but can't be used by eac3to until madshi can parse the track and add new libav dll's.

By the moment you can do, for instance:

ffmpeg -i 000000.m2ts -map 0:1 recoded.flac
Thanks for this, tho what I would really like to do is recode to AC3 640 kbps. Is this possible?
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Old 29th September 2014, 10:08   #12824  |  Link
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Quote:
Originally Posted by JimmyBarnes View Post
Thanks for this, tho what I would really like to do is recode to AC3 640 kbps. Is this possible?
Of course you can recode to AC3 with ffmpeg, but a 7.1 must be downmix to 5.1 and is always better extract the AC3 included in the track with tsMuxeR (like I say in a precedent post).

BTW:

ffmpeg -i 00000.m2ts -map 0:1 -acodec ac3 -ac 6 -ab 640000 recoded.ac3
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Old 29th September 2014, 16:08   #12825  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Of course you can recode to AC3 with ffmpeg, but a 7.1 must be downmix to 5.1 and is always better extract the AC3 included in the track with tsMuxeR (like I say in a precedent post).

BTW:

ffmpeg -i 00000.m2ts -map 0:1 -acodec ac3 -ac 6 -ab 640000 recoded.ac3
Thanks for the above.

Does this actually convert the entire THD+AC3 to AC3 (rather than simply extract the AC3 stream, as you suggest with tsMuxeR)?
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Old 29th September 2014, 17:01   #12826  |  Link
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Quote:
Originally Posted by JimmyBarnes View Post
Could someone indicate how is this supposed to be used, especially regarding using eac3to convert the Age of Extinction THD+AC3 stream to DTS or AC3.

TIA
I simply used this:

ffmpeg -i "test.m2ts" -vn -ac 8 -acodec pcm_s24le -f wav c:\video\temp.wav
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Old 29th September 2014, 19:44   #12827  |  Link
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Quote:
Originally Posted by JimmyBarnes View Post
Thanks for the above.

Does this actually convert the entire THD+AC3 to AC3 (rather than simply extract the AC3 stream, as you suggest with tsMuxeR)?
Re-read what you quoted.
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Old 30th September 2014, 02:51   #12828  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
ffmpeg -i 00000.m2ts -map 0:1 -acodec ac3 -ac 6 -ab 640000 recoded.ac3
This worked and the resulting AC3 synced OK with the recoded x264 video
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Old 30th September 2014, 11:57   #12829  |  Link
minhjirachi
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Waiting for an update of eac3to.
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Old 30th September 2014, 12:11   #12830  |  Link
Lenmaer
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We were promised updates for over a year now, will it finally happen?
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Old 30th September 2014, 12:18   #12831  |  Link
JimmyBarnes
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Quote:
Originally Posted by minhjirachi View Post
Waiting for an update of eac3to.
This would be the ideal solution to the Dolby Atmos issue, I would expect..

But the ffmpeg solution described here can be used meanwhile.
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Old 5th October 2014, 00:03   #12832  |  Link
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Hi could you explain me the difference between 16bits and 24bits DTS ? And the real utility to use -dontPatchDts option or not ? Thanks !
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Old 5th October 2014, 03:27   #12833  |  Link
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In the standard (lossy) DTS headers there are a field to remember the bitdepth (precission) of the source PCM used to encode the DTS.
After the encode this info is useless because we can't recover the original precission.

By default eac3to change this field to put always 24 bits to remember than any lossy format (AC3, DTS, ...) must be decoded at last to 24 bits, even if the source PCM was 16 bits, to preserve the max quality.
Then use -dontPatchDts is useless because don't change any relevant info in the stream.
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Old 5th October 2014, 10:49   #12834  |  Link
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Thanks for your response. So it's useless now to use this option because any player can read a "virrtual" 24bits stream ? Even in a MKV ?
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Old 5th October 2014, 13:31   #12835  |  Link
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Players don't use that field because lossy formats don't have bitdepth, must be ignored in standalone DTS or in any container.
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Old 5th October 2014, 15:31   #12836  |  Link
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Ok thanks for these informations !
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Old 19th October 2014, 16:47   #12837  |  Link
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Format is not supported

Quote:
Originally Posted by RoseM View Post
Hello

Someone could see why the audio of the file below isn't recognized by the program?

http://www.mediafire.com/download/5w...Globo_HD_SP.ts

Thanks
Quote:
Originally Posted by Atak_Snajpera View Post
Because this format is not supported by eac3to tool.

Code:
Audio #1
ID                                       : 274 (0x112)
Menu ID                                  : 59200 (0xE740)
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : HE-AAC / LC
Muxing mode                              : LATM
Codec ID                                 : 17
Duration                                 : 1mn 11s
Bit rate mode                            : Variable
Channel(s)                               : 2 channels
Channel positions                        : Front: L R
Sampling rate                            : 48.0 KHz / 24.0 KHz
Compression mode                         : Lossy
Delay relative to video                  : -400ms
Could add support for this format in a update future .

thanks
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Old 19th October 2014, 19:36   #12838  |  Link
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That is not format audio, is muxing mode. The audio format for you code is AAC, and eac3to already support that format.
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Old 19th October 2014, 22:48   #12839  |  Link
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The message from eac3to about the aac tracks of this file is "parsing failed".
But it works when put in MKA (and probably also MKV), eac3to has maybe problems with aac in TS container.

Quote:
Originally Posted by RoseM View Post
Could add support for this format in a update future .
Open your TS in MKVMerge, uncheck video if not needed, export in mka, open the mka in eac3to.
Strange thing : the aac 2.0 becomes 3.0 when in mka and the frequency becomes 22.05 KHz ;

Code:
Audio
ID                                       : 1
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : LC
Codec ID                                 : A_AAC
Duration                                 : 1mn 8s
Channel(s)                               : 3 channels
Channel positions                        : Front: L C R
Sampling rate                            : 22.05 KHz
Compression mode                         : Lossy
Default                                  : Yes
Forced                                   : No

edit : I spoke too quickly ; analyzing works but not decoding ;

Code:
eac3to v3.27
command line: "C:\Program Files\eac3to\eac3to.exe" "F:\09-30_12-25-28_Globo HD_SP.mka" "F:\09-30_12-25-28_Globo HD_SP.wavs"
------------------------------------------------------------------------------
MKA, 1 audio track, 0:01:09
1: AAC, 3.0 channels, 22kHz
Track 1 is used for destination file "09-30_12-25-28_Globo HD_SP.wavs".
[a01] Extracting audio track number 1...
[a01] Decoding with DirectShow (Nero Audio Decoder 2)...
[a01] Getting "Nero Audio Decoder 2" instance failed.  <ERROR>
Aborted at file position 220599.  <ERROR>

Last edited by Music Fan; 19th October 2014 at 23:09.
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Old 20th October 2014, 07:35   #12840  |  Link
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@RoseM

1) Extract with eac3to (ignore warnings)

2) Use vlc to convert LATM to standard .m4a (see http://forum.doom9.org/showthread.ph...84#post1453484 ), for instance:

vlc.exe -I dummy aud_latm.aac --sout #std{access=file,mux=mp4,dst="aud_nolatm.mp4"} vlc://quit
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