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Old 21st October 2011, 07:05   #1  |  Link
xekon
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Guide AviSynth encode x264 trim splice cut commercials autocrop srt subtitle sync

This guide shows how to use AviSynth to encode x264, and cut out commercials in the process.

You may need to adjust the paths or rename files to reflect where you installed AviSynth, I installed AviSynth directly to my 'E:\' partition eg:
E:\AviSynth
E:\AviSynth\plugins

My batch files are directly in my AviSynth folder:
E:\AviSynth\bd.bat
E:\AviSynth\bd-go.bat

software required:
Code:
AviSynth 2.6
ffms2.dll    (put in AvisSynth plugins folder)
NicAudio.dll (put in AvisSynth plugins folder)
AutoCrop.dll (put in AvisSynth plugins folder)
ffmpeg       (E:\AviSynth\ffmpeg\bin\ffmpeg.exe)
mkvtoolnix   (E:\AviSynth\mkvtoolnix\mkvmerge.exe)
x264         (E:\AviSynth\x264.exe)
avs2pipe     (E:\AviSynth\avs2pipe.exe) http://forum.doom9.org/showthread.php?p=1527344
trimsrt      (E:\AviSynth\trimsrt.exe) http://forum.doom9.org/showthread.php?t=162828

neroAacEnc   (E:\AviSynth\neroAacEnc.exe)
*or*
qaac         (E:\AviSynth\qaac.exe)
If you choose qaac, it requires QuickTime to be installed too
There are also some codecs that are required, but if you install CCCP http://www.cccp-project.net/ it will take care of them all.


bd.bat <in-file> <fps> <audio-stream-id> <audio-channels> <out-file> <frame1A> <frame1B> <frame2A> <frame2B>
example:
bd.bat E:\yyh\BDMV\STREAM\00039.m2ts 23.976 0:2 6 004 2832 16675 16872 31270

The above line will take me from the 00039.m2ts file to a trimmed, spliced, cropped, and x264 encoded 004.mkv file with Quicktime TVBR AAC encoded audio (or neroaac), and a trim and adjusted .srt subtitle file if applicable.

to determine which audio stream you want and how many channels it has use ffmpeg (for 5.1 use 6, for stereo use 2):
Code:
E:\AviSynth\ffmpeg\bin\ffmpeg -i E:\yyh\BDMV\STREAM\00036.m2ts
bd.bat:
Code:
SET FPS=%~2
if "%~2"=="23.976" SET FPS=24000/1001
del bd.avs
echo A = RaWavSource("E:\AviSynth\job.wav")>>bd.avs
echo V = FFVideoSource("E:\AviSynth\job.mkv")>>bd.avs
echo AudioDub(V, A)>>bd.avs
if not "%~6"=="" echo AlignedSplice(Trim(%6, %7), Trim(%8, %9))>>bd.avs
echo AutoCrop(0,2,2,0,0,0,0,30,5,0,-1,0)>>bd.avs
E:\AviSynth\ffmpeg\bin\ffmpeg -i %1 -vn -sn -y -map %3 -drc_scale 0 -ac %4 -async 1 -acodec pcm_s24le -f wav job.wav
E:\AviSynth\ffmpeg\bin\ffmpeg -i %1 -an -sn -y -vcodec copy -f rawvideo job.h264
E:\AviSynth\mkvtoolnix\mkvmerge -o E:\AviSynth\job.mkv --engage keep_bitstream_ar_info --default-duration 0:%FPS%fps --compression 0:none -d 0 -A -S -T --no-global-tags --no-chapters E:\AviSynth\job.h264
E:\AviSynth\avs2pipe -wav bd.avs | qaac --tvbr 90 --quality 2 --rate keep --ignorelength - -o job.m4a
::E:\AviSynth\avs2pipe -wav bd.avs | neroAacEnc.exe -q 0.5 -ignorelength -if - -of job.m4a
E:\AviSynth\x264 --preset veryslow --crf 22.0 --ref 5 --bframes 16 --no-fast-pskip --output "jobz.mkv" "E:\AviSynth\bd.avs"
E:\AviSynth\mkvtoolnix\mkvmerge -o E:\AviSynth\%5.mkv --engage keep_bitstream_ar_info --compression 1:none -d 1 -A -S -T --no-global-tags --no-chapters E:\AviSynth\jobz.mkv --compression 1:none -a 1 -D -S -T --no-global-tags --no-chapters E:\AviSynth\job.m4a
::E:\AviSynth\trimsrt -i %5.srt -f %FPS% -t %6,%7,%8,%9
::E:\AviSynth\mkvtoolnix\mkvmerge -o E:\AviSynth\%5.mkv --engage keep_bitstream_ar_info --compression 1:none -d 1 -A -S -T --no-global-tags --no-chapters E:\AviSynth\jobz.mkv --compression 1:none -a 1 -D -S -T --no-global-tags --no-chapters E:\AviSynth\job.m4a --compression 0:none -s 0 -D -A -T --no-global-tags --no-chapters E:\AviSynth\%5new.srt
adjust your --ref according to your source and desired level compatibility: http://forum.handbrake.fr/download/f...d=49&mode=view
I use --crf 22.0 for Bluray and --crf 20.0 for DVD, some people will use lower or higher, you have to figure out what works for you: https://trac.handbrake.fr/wiki/ConstantQuality

To make use of trimsrt uncomment the last two lines in bd.bat by removing the two :: from the beginning of that line, and comment out the mkvmerge line before the last two lines

To use trimsrt you have to have the .srt file in the same output directory as the other files (job.mkv, job.wav, bd.bat). Also the .srt file has to have the same filename as your output filename for the finished mkv

bd.bat E:\yyh\BDMV\STREAM\00039.m2ts 004 2832 16675 16872 31270
(in this example the output file name is 004, so you would need to name the .srt file 004.srt prior to running bd.bat or bd-go.bat, which means you have to prepare your subs first)

With the bd.bat batch file I can queue up as many jobs as I want, by making another batch file that calls bd.bat eg:

bd-go.bat:
Code:
CALL bd.bat E:\yyh\BDMV\STREAM\00036.m2ts 23.976 0:2 6 001 2112 17685 17854 30940
CALL bd.bat E:\yyh\BDMV\STREAM\00037.m2ts 23.976 0:2 6 002 2833 16127 16296 31296
CALL bd.bat E:\yyh\BDMV\STREAM\00038.m2ts 23.976 0:2 6 003 2832 16463 16632 31295
CALL bd.bat E:\yyh\BDMV\STREAM\00039.m2ts 23.976 0:2 6 004 2832 16703 16872 31295
CALL bd.bat E:\yyh\BDMV\STREAM\00040.m2ts 23.976 0:2 6 005 2856 17471 17640 31295
CALL bd.bat E:\yyh\BDMV\STREAM\00041.m2ts 23.976 0:2 6 006 2832 16727 16896 31295
CALL bd.bat E:\yyh\BDMV\STREAM\00042.m2ts 23.976 0:2 6 007 2856 16367 16536 31295
CALL bd.bat E:\yyh\BDMV\STREAM\00043.m2ts 23.976 0:2 6 008 2856 15935 16104 31295
The reason for trimming and using the 2 different frame ranges is to cut out the commercial, here is an explanation of the line:

CALL bd.bat <in-file> <fps> <audio-stream-id> <audio-channels> <out-file> <frame1A> <frame1B> <frame2A> <frame2B>

<frame1A> <frame1B> is the frame range to keep before the commercial and,
<frame2A> <frame2B> is the frame range to keep after the commercial

To figure out the frame range that I needed I used Avidemux, but you could also use AvsP: http://avisynth.org/qwerpoi/ (it has a trim editor with a preview window so you can see exactly what you will get)

The Frame Range is optional, if you would like to encode the entire file you can (although if you dont need the frame ranges there are easier ways to do all this)

Last edited by xekon; 27th October 2011 at 07:03.
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Old 21st October 2011, 15:49   #2  |  Link
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- If you use:
...
echo A = RaWavSource("E:\AviSynth\job.wav")>>bd.avs
...
E:\AviSynth\eac3to\eac3to %1 2: job.mkv 3: job.wav
...

This work for all audio input: pcm, dts, dtshd, dtsma, truehd, ac3, eac3
(RaWavSource included in NicAudio.dll)

- For what: mkvmerge.exe ... --aac-is-sbr 1:1 ?

When you encode with
neroAacEnc.exe -q 0.5
the output is always AAC-LC, without SBR part.
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Old 21st October 2011, 18:50   #3  |  Link
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you have to be very careful when trim...often commercial audio is 2.0 ac3 and the tv show is 5.1 ac3...even one 2.0 ac3 frame will ruin the audio...anyway some tv shows have the "previously on" part as 2.0 ac3...
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Last edited by b66pak; 21st October 2011 at 19:10.
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Old 21st October 2011, 20:38   #4  |  Link
xekon
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Quote:
Originally Posted by tebasuna51 View Post
When you encode with
neroAacEnc.exe -q 0.5
the output is always AAC-LC, without SBR part.
I wondered about that with nero, thanks I have edited the top post.

When you use eac3to to go from one format to another like .wav does it do so without quality loss? ( IIRC eac3to to .wav outputs a stereo/2 channel wav file right? so you would lose the 5.1 audio)

Quote:
Originally Posted by b66pak View Post
you have to be very careful when trim...often commercial audio is 2.0 ac3 and the tv show is 5.1 ac3...even one 2.0 ac3 frame will ruin the audio...anyway some tv shows have the "previously on" part as 2.0 ac3...
_
Very good to know.

running the command below shows all streams for a file:
eac3to.exe E:\yyh\BDMV\STREAM\00039.m2ts

For the tv shows that you are talking about, this would show if there was a 2.0 ac3 stream correct?

Is there a method you use before trimming to check if there is a segment that uses 2.0 ac3?

Also how do you go about making sure your trim avoids the 2.0 ac3? do you just listen to the quality to determine, or is there a more definitive test.

Last edited by xekon; 22nd October 2011 at 05:56.
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Old 21st October 2011, 20:54   #5  |  Link
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you have to demux the .ac3 and check it with delaycut...or use ffmpeg for decoding:

Code:
ffmpeg -i tvshow.m2ts -vn -sn -drc_scale 0 -ac 6 -acodec pcm_s24le -f wav tvshow.wav -async 1
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Last edited by b66pak; 21st October 2011 at 21:00.
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Old 22nd October 2011, 00:16   #6  |  Link
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Thanks for the reply!

I changed the batch file so that it avoids the issue (I have never ran into the problem you describe, but I have not ripped and encoded many series yet.)

I no longer decode/demux using eac3to, using ffmpeg now instead.
the avs uses the ffmpeg decoded audio file, which gets piped into qaac/neroaac

Last edited by xekon; 27th October 2011 at 07:10.
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Old 22nd October 2011, 19:04   #7  |  Link
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it's ok...if the m2ts have multiple audio tracks you must add "-map int:int" to the ffmpeg line...
to determine the values run:
Code:
ffmpeg -i tvshow.m2ts
you will get something like this:
Quote:
Input #0, mpegts, from 'bbc_one_hd.1080i.ts':
Duration: 00:00:33.70, start: 3.000000, bitrate: 12129 kb/s
Program 6941
Stream #0:0[0x1518]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0:1[0x151a]: Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16, 256 kb/s
Stream #0:2[0x1519]: Audio: ac3 ([6][0][0][0] / 0x0006), 48000 Hz, stereo, s16, 192 kb/s
Stream #0:3[0x151b](eng): Subtitle: dvb_teletext ([6][0][0][0] / 0x0006)
Stream #0:4[0xf05]: Unknown: none ([11][0][0][0] / 0x000B)
Stream #0:5[0xf06]: Unknown: none ([11][0][0][0] / 0x000B)
Stream #0:6[0xf07]: Unknown: none ([11][0][0][0] / 0x000B)
Stream #0:7[0xf08]: Unknown: none ([11][0][0][0] / 0x000B)
Stream #0:8[0x151c](eng): Subtitle: dvb_subtitle ([6][0][0][0] / 0x0006)
Stream #0:9[0xf00]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:10[0xf01]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:11[0xf02]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:12[0xf03]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:13[0xf04]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:14[0x901]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:15[0x902]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:16[0x903]: Unknown: none ([5][0][0][0] / 0x0005)
Stream #0:17[0x914]: Unknown: none ([5][0][0][0] / 0x0005)
if you want the .ac3 the line should look like this:
Code:
ffmpeg -i %1 -vn -sn -drc_scale 0 -ac 6 -acodec pcm_s24le -f wav -map 0:2 -y job.wav -async 1
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Last edited by b66pak; 22nd October 2011 at 19:19.
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Old 22nd October 2011, 19:27   #8  |  Link
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Yes, I read about -map. if you don't use it then ffmpeg uses the first audio track. What do you think of the new .batch file?

it uses RaWavSource() like tebasuna51 suggested, and it uses ffmpeg like you suggested, instead of using eac3to, i also added the -y option so that it overrides any existing file, this is good because the batch file can be used to queue more than one job, this way I do not have to delete the intermediate file at the end of a file/episode, it can just be overwritten.

Now using AlignedSplice(Trim(%3, %4), Trim(%5, %6)) should not be a problem even if one or more frames originally had ac 2.0.

Last edited by xekon; 22nd October 2011 at 23:23.
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Old 22nd October 2011, 19:47   #9  |  Link
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you should mod it to work with:

Code:
CALL bd.bat E:\yyh\BDMV\STREAM\00039.m2ts 25 2 3 004 2832 16675 16872 31270
CALL bd.bat <input-file> <video-fps> <video-stream-id> <audio-stream-id> <output-file-name> <frame1A> <frame1B> <frame2A> <frame2B>
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Old 22nd October 2011, 19:51   #10  |  Link
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Thanks for the reply, I was thinking the same thing, because if you are using the script in queue mode and have files with different stream IDs it could be a problem. I will edit it now.
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Old 22nd October 2011, 19:55   #11  |  Link
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don't forget the fps (this is for trimsrt)...
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Old 22nd October 2011, 23:00   #12  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
- If you use:
...
echo A = RaWavSource("E:\AviSynth\job.wav")>>bd.avs
...
(RaWavSource included in NicAudio.dll)
I was reading a post tebasuna51 made last year: https://swesub.nu/showthread.php?p=1454464#post1454464

it says that nicaudio RaWavSource() uses 32bit float.

If I am going to use nicaudio's RaWavSource() in my avs, for audio decoded using ffmpeg, would it be best to decode with ffmpeg to the same type?

Code:
E:\AviSynth\ffmpeg\bin\ffmpeg -i 00037.m2ts -vn -sn -y -drc_scale 0 -ac 6 -async 1 -acodec pcm_s24le -f wav job2.wav
-acodec pcm_s24le gives me PCM signed 24-bit little-endian

so I am not sure that is the best way to do it.


ffmpeg supports many flavors of 32bit:
s32be, s32le, u32be, u32le, f32be, f32le

assuming I want to keep it at 32 bit float so that it is the same as RaWavSource() that narrows it down to: f32be, f32le

I am still unclear as to what would be the best -acodec to use if I am going to be using RaWavSource() in my avs.

my best guess is:
Code:
E:\AviSynth\ffmpeg\bin\ffmpeg -i E:\yyh\BDMV\STREAM\00037.m2ts -vn -sn -y -map 0:2 -drc_scale 0 -ac 6 -async 1 -acodec pcm_f32le -f wav job3.wav
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Old 23rd October 2011, 00:24   #13  |  Link
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I don't know how work now ffmpeg. Long time ago the decoded data was always at 16 bits only, of course the output can be upsize until float 32 but the precision was always 16 bits.

For audio quality of TV sources I think is enough 24 bits, but RaWavSource and NeroAacEnc support also 32 bit float (the mantissa is 24 bits, then the precision is the same than 24 bits int)
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Old 23rd October 2011, 08:05   #14  |  Link
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Quote:
http://avisynth.org/mediawiki/Intern...trol_functions
OPT_AllowFloatAudio | v2.57 | global OPT_AllowFloatAudio = True
This option enables WAVE_FORMAT_IEEE_FLOAT audio output. The default is to autoconvert Float audio to 16 bit.
It will be changed into 16bit by avisynth itself if you pass float to the tool using vfw interface(e.g. ffmpeg, wavi, etc) without ”global OPT_AllowFloatAudio = True".
And generaly, you should avoid use of global variables if possible.
Thus, avs2pipe(mod) is better choice than wavi if you want to use float audio.
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Old 23rd October 2011, 08:14   #15  |  Link
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thanks for the reply, I did not even know about avs2pipe. I will be checking it out for sure!
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Old 23rd October 2011, 09:48   #16  |  Link
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Quote:
Originally Posted by Chikuzen View Post
It will be changed into 16bit by avisynth itself if you pass float to the tool using vfw interface(e.g. ffmpeg, wavi, etc) without ”global OPT_AllowFloatAudio = True".
And generaly, you should avoid use of global variables if possible.
Actually, it doesn't have to be global - you can set it as a normal variable at the outer level of your script. You only have to make it global if you are setting it inside a function.
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Old 23rd October 2011, 10:01   #17  |  Link
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Quote:
Originally Posted by Gavino View Post
Actually, it doesn't have to be global - you can set it as a normal variable at the outer level of your script. You only have to make it global if you are setting it inside a function.
You are correct. it was my misapprehension.
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Old 27th October 2011, 07:06   #18  |  Link
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Thank you everyone for the wonderful feedback and ideas.

I have made a few changes so far, and incorporated many of your ideas.

I have updated the top post to the updated version of the batch script.
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Old 27th October 2011, 18:02   #19  |  Link
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this line is not OK:
Quote:
E:\AviSynth\ffmpeg\bin\ffmpeg -i %1 -an -sn -y -vcodec copy -f rawvideo job.h264
it should be (or else mkvmerge will reject it!):
Quote:
E:\AviSynth\ffmpeg\bin\ffmpeg -i %1 -an -sn -y -vcodec copy -bsf h264_mp4toannexb -f rawvideo job.h264
i don't see why do you need to demux the source to raw and then remux it to mkv (with mkvmerge)...you can do this in one step with eac3to...even better you leave it as it is and decode it with:
Quote:
echo V = FFVideoSource("E:\AviSynth\%1")>>bd.avs
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Old 27th October 2011, 20:09   #20  |  Link
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I have tested the script with a couple of sources start to finish, mkvmerge did not have a problem with this line for me:
Code:
E:\AviSynth\ffmpeg\bin\ffmpeg -i %1 -an -sn -y -vcodec copy -f rawvideo job.h264
I wanted to find a good explanation of when 'h264_mp4toannexb' should be used, it is only mentioned once on this page: http://ffmpeg.org/ffmpeg.html
Quote:
ffmpeg -i h264.mp4 -c:v copy -vbsf h264_mp4toannexb -an out.h264
and in this example it is going from a h264 stream in a .mp4 file, maybe I am not having the issue because my h264 stream comes from a .m2ts file?

What I will probably do is edit the batch file to detect if the input file ends in .mp4, and if it does then add "-vbsf h264_mp4toannexb" to the ffmpeg video demux line.

HOWEVER, I moved away from eac3to, because for some odd reason when I used it, some of the episodes where just Blank/Black and Avisynth through an error.

but If I use mkvmerge to pack it into mkv then I no longer have the blank/black video error.

The reason I am not using:
Code:
echo V = FFVideoSource("E:\AviSynth\%1")>>bd.avs
is because it could possibly create issues with ffms2, whereas the mkv should have no issues at all:
http://ffmpegsource.googlecode.com/s...-avisynth.html
Quote:
M2TS, TS: Seeking seems to be off a few frames here and there
Basically, FFMS2 might be able to open .m2ts files on its own but it'll probably have somewhat inaccurate seeking like it does with .ts files, which is why I demux then mux to mkv.

Last edited by xekon; 27th October 2011 at 21:05.
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