Audio adjustment for 25fps to 24fps

[ranosb]

I have a mkv hdtv video that was encoded at 25fps but will not play on my Sony Bdp-s301 when I convert the video to Avchd.
So I adjust the settings in Tsmuxer to change the frames p/s to 24 and it now plays but the audio is out of sync.
What adjustment to the audio do I need to sync this video going from 25fps to 24, or even 23.967fps?
Thanks experts!

Format                           : Matroska
File size                        : 1.09 GiB
Duration                         : 39mn 28s
Overall bit rate                 : 3 963 Kbps
Encoded date                     : UTC 2009-02-09 12:46:53
Writing application              : mkvmerge v2.4.1 ('Use Me') built on Dec  5 2008 18:30:05
Writing library                  : libebml v0.7.7 + libmatroska v0.8.1

Video
ID                               : 1
Format                           : AVC
Format/Info                      : Advanced Video Codec
Format profile                   : High@L3.1
Format settings, CABAC           : Yes
Format settings, ReFrames        : 3 frames
Muxing mode                      : Container profile=Unknown@3.1
Codec ID                         : V_MPEG4/ISO/AVC
Duration                         : 39mn 28s
Bit rate                         : 3 500 Kbps
Nominal bit rate                 : 3 576 Kbps
Width                            : 1 280 pixels
Height                           : 720 pixels
Display aspect ratio             : 16/9
Frame rate                       : 25.000 fps
Resolution                       : 24 bits
Colorimetry                      : 4:2:0
Scan type                        : Progressive
Bits/(Pixel*Frame)               : 0.152
Stream size                      : 988 MiB (88%)
Writing library                  : x264 core 65 r1046 71d34b4
Language                         : English

Audio
ID                               : 2
Format                           : AC-3
Format/Info                      : Audio Coding 3
Codec ID                         : A_AC3
Duration                         : 39mn 28s
Bit rate mode                    : Constant
Bit rate                         : 384 Kbps
Channel(s)                       : 6 channels
Channel positions                : Front: L C R, Surround: L R, LFE
Sampling rate                    : 48.0 KHz
Stream size                      : 108 MiB (10%)

[johns0]

Use yr_eac3to_more_gui and add the ac3 extracted and in the eac3to audio tab select pal speedup/slowdown 25>24/1.001,also adjust the other settings to ac3 for output and name ect.

I found in a few mkv that i converted to ntsc from pal had to be stretched to the exact time length as the video because converting it the way i described didnt give the exact length.Also use 24/1.001 in tsmuxer to change the video fps as it will give you the proper 23.976 fps,not 24.

[ranosb]

Thanks for the reply, yr_eac3to_more_gui seems user unfriendly.
I set perimeters then click "run command line"? and it clears all the settings after every run?!

[johns0]

You also have to choose input files or folder,select save type as ac3,file name then it will run,its not that hard to figure out.

[ranosb]

Well I know now why it wouldn't work, if you set Quality to 4 on the audio, it doesn't run...
I took the 1gb source file and cut it to a 50mb Mkv for testing purposes.
Used Tsmuxer to demux video and audio,
Ran Eac3 Gui and imported the ac3 and used the perimeter u suggested, Yes 24/1.001 is 23.967
Tsmuxer again joining the new ac3 and the org video using 24/1.001 output to Avchd.

IT WORKS! but this is a small 2min file. I will run the full Mkv as same and see if the audio STAYS in sync to the end.

Nice job and reply...
Seems if Tsmuxer added a 24/1.001 to the audio options instead of delay in ms, this could all be done w/o demuxing...

Eac3 Gui settings can't be saved?

It defaulted to 640kbps on the audio, had set to same as source of 384kbps.
While running, message "Clipping detected, a 2nd pass will be necessary..." came up.

[ranosb]

RESULT: The 50mb file worked but not the entire video, seems the audio was silent?
Did the 2nd pass on the audio destroy something?

Original audio file:

Format                           : AC-3
Format/Info                      : Audio Coding 3
FileSize/String                  : 108 MiB
Duration/String                  : 39mn 25s
OverallBitRate/String            : 384 Kbps

Audio
Format                           : AC-3
Format/Info                      : Audio Coding 3
Duration/String                  : 39mn 25s
BitRate_Mode/String              : Constant
BitRate/String                   : 384 Kbps
Channel(s)/String                : 6 channels
ChannelPositions                 : Front: L C R, Surround: L R, LFE
SamplingRate/String              : 48.0 KHz
StreamSize/String                : 108 MiB (100%)

New audio file:

Format                           : AC-3
Format/Info                      : Audio Coding 3
FileSize/String                  : 113 MiB
Duration/String                  : 41mn 6s
OverallBitRate/String            : 384 Kbps

Audio
Format                           : AC-3
Format/Info                      : Audio Coding 3
Duration/String                  : 41mn 6s
BitRate_Mode/String              : Constant
BitRate/String                   : 384 Kbps
Channel(s)/String                : 6 channels
ChannelPositions                 : Front: L C R, Surround: L R, LFE
SamplingRate/String              : 48.0 KHz
StreamSize/String                : 113 MiB (100%)

[johns0]

Just use the unedited ac3 and run that thru eac3to and mux with the unedited video.

[ranosb]

How would I do that? Seems every tab is a different function, if you switch between tabs, selections are lost.
If you run the program, when its done(Error or not) everything is lost/reset.
Very much a hassle to use...

[johns0]

Nothings lost when you switch between tabs in eac3to,just load the ac3 in the first tab and then enter the settings in the second tab-file name,save as type ac3,input file or folder,pal speedup/slowdown slowdown 25->24/1.001.

You can also save some of the settings in the preference tab.

[manono]

You can also save 6 Mono WAV files in BeSweet, at the same time using a PAL2NTSC Preset. Then reencode the WAV files back to AC3 using Aften or somesuch.

[ranosb]

Ok, using Besweet, It worked in ac3 mode but, the file converted 5.1 to 2ch stereo.?
I have the "create 5.1 Ac3" checkbox selected under AC3/OGG/PCM, what am I doing wrong?
Thanks!

[manono]

But that wasn't my suggestion, so you're on your own.

[ranosb]

Using Besweet creating 6 mono wav files then muxing with Aften like you suggested, worked!
Audio sync was right on all the way to the last minutes of video.

There was an audio pop, just as the video started and randomly there are sounds like a cash register being closed, NOT GOOD.

Using Aften Mux Wizard, I noticed the +LFE.wav(Subwoofer) was defaulted as not selected, (I added it) any reason for this?

Only thing left is getting the ac3 to produce 5.1 ch instead of 2ch...Maybe aften is causing the audio artifacts...

[manono]

I mentioned Aften because it's known to be a good freeware AC3 encoder. I use a different (commercial) program for encoding for AC3, so I can't give any specific advice about any problems it might have created. Maybe there are other programs you could try. Maybe someone more familiar with Aften can offer advice.

[ranosb]

Thanks for your replies.
Yes Aften was the problem, since I now fiqured out how to use Besweet (ac3 25fps to ac3 23.967) keeping the 5.1 audio source and it came out perfect!

It uses 90-100% Cpu power though, and has crashed a few times on my single core 2GHZ CPU, so im going to move over to the Dual core 2.6Mhz and see how it performs on the notebook...

BTW, what commerical software are you using in place of Besweet?

[manono]

I use BeSweet to change the 'framerate' and make WAV files from AC3 almost daily. For turning those WAV files back into AC3 I use Soft Encode. I (and many others) have never been particularly impressed with the AC3 audio BeSweet creates. You might open a WAV file BeSweet created in a WAV Editor such as Audacity and then compare it with a WAV created from the AC3 made by BeSweet.

[ranosb]

Is the demonstration download of Soft Encode functional for just 6ch wav to ac3, like what youv'e been doing?

[manono]

Demonstration download? I don't know anything about a demo version and what it can or cannot do. Sorry.

[ranosb]

The demo is no longer available because it has been discontinued...
Have any other recommendations for commercial software to do the job?

[manono]

No, sorry.

[lordsmurf]

For stereo, yes, for multi-channel, no.

[ranosb]

BeSweet seems to work fine, for the audio, but how about subtitles when you change the frame rate?

[manono]

...but how about subtitles when you change the frame rate?

What about them? Change their framerate as well. Maybe if you told us the sub format we could tell you how to accomplish it.

[ranosb]

Subtitles with the .sup extension...

[manono]

There are 2 kinds. If SUP from a DVD, then I use SubtitleCreator to do framerate conversions. It's in something like Synchronize->Framerate Conversion. Something like that. I'm at a different computer at the moment.

If it's the SUP files from a Blu-Ray disc (as I suspect), then BDSup2Sub is supposed to be able to do it. Here's the thread on it:

http://forum.doom9.org/showthread.php?t=145277

I've never used it, so if you have any questions on its use someone else will have to help.

[ranosb]

Is the demonstration download of Soft Encode functional for just 6ch wav to ac3, like what youv'e been doing?

Got a hold of Soft Encode 1.0 B19, took 85min to convert 6ch wavs to Ac-3, is the sound quality that much better than just using Besweet Ac-3 to Ac-3?

Ever heard of using Audacity ac-3, bypassing Besweet altogether? It also has pitch adjust change, which im not sure about with Besweet...

[ranosb]

Where do you get the azid file for transcoding DTS in besweet?

[manono]

Where do you get the azid file for transcoding DTS in besweet?

http://forum.doom9.org/showthread.php?s=&threadid=68300#post424069

[ranosb]

Awesome guide, Thank You!

[tsuyoshi85]

It also has pitch adjust change, which im not sure about with Besweet...

Pitch is never to be changed in a PAL to film framerate conversion since PAL material was just sped up and not pitch-altered. As manono says, BeSweet for decoding and speed change, Soft Encode (be sure to encode your ac-3 files in big endian mode) for re-encoding. BeSweet created ac-3s' volume is ridiculously low.