Normalizing Multiple Audio

[hello_hello]

@hello_hello

I wonder if there is some good GUI for FFMPEG to use your options.

Off the top of my head there's AVANTI, FFQueue and WinFF.

I've tried AVANTI briefly a few times and ended up suicidal, but that doesn't necessarily mean it's a bad program. Maybe I just haven't got my head around it yet.

WinFF uses a preset system which allows you to create your own, and in theory it should be easy, but in practice I couldn't get it to work with the command line you've been using. It creates a batch file for encoding but it re-writes some of the command line options in the process and I couldn't stop it from getting it wrong. Maybe I'm missing something but it kept messing with the -vf parameters. If I put them after specifying the video codec in the command line it deleted -vf from the command line and if I put -vf first it added a comma after the filters that turned out to be a showstopper.
It's got a section for specifying additional command line parameters independent of the preset used but I couldn't get that to work properly either. Whenever I tried to use it, WinFF gave every file the same output name so batch encoding becoming useless. Maybe you'll have better luck. It's a pity I didn't because WinFF looks like it should be pretty straight forward.

I haven't played with FFQueue yet. It works using a similar preset system to WinFF, but while WinFF is pretty much all command line, FFQueue forces you to select some options in the GUI, and it seems video and audio filter presets need to be created so they can be added to a conversion preset, but until a preset is saved and a file loaded I don't think there's any way to know what command line FFQueue will create so I think creating a conversion preset that does what you want it to do might involve a fair bit of trial and error.
It's odd because FFQueue lets you bypass the preset system and specify a command line when loading a single file for encoding, but batch mode forces you to use a conversion preset. I don't know why.

Maybe someone can suggest a different ffmpeg GUI.

[hello_hello]

Isn't it always the way..... I spent a fair bit of time messing about trying to get WinFF to work, only to discover AnotherGUI later on, and it took no time at all.

You might need to click on the Executables button to tell it where to find ffmpeg, and it seems to assume ffprobe and ffplay are in the same location. Creating a preset was very easy, and works as expected. Adding the following to the "first pass" section when creating a new preset does the trick.

-i "<SourceFileName>" -vf "fspp=5,setdar=384/288,scale=384:288" -r "25" -c:v mpeg4 -profile:v mpeg4_asp -slices 1 -q:v 4 -mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -vtag DIVX -c:a libmp3lame -q:a 5 -af "aformat=sample_fmts=fltp,highpass=f=120,pan=stereo|FL < FL + 1.414FC + .5BL + .5SL|FR < FR + 1.414FC + .5BR + .5SR,aresample=resampler=soxr:osr=48000:dither_method=0,dynaudnorm=m=50.0:b=1:s=25.0:p=0.5,firequalizer=gain='if(lte(f,10000),0,-INF)'" "<OutputPath><OutputFileName>.avi"

(I changed the value for -slices to an actual number, so change it to whatever you prefer)

Under Preferences there's an option for the maximum number of simultaneous processes, which seems to mean the maximum number of simultaneous jobs it'll run. It defaulted to 4 for me.

Other than that it should just be a matter of loading the source files with the appropriate preset selected, choosing an output location and clicking "Go". Getting it to work was pretty easy.

[ConverterCrazy]

@hello_hello

Thanks.

Sorry for the delay. I was trying many options.

Tried AnotherGUI and it stopped working as has happened during running your scripts for FFMPEG.


I found My FFVideoconverter & Editor here: https://sourceforge.net/projects/myffvideoconver/?source=typ_redirect

Works real fast, excellent audio and video quality with proper settings.

But, the sad part is: the defects in audio, as you pointed out in earlier posts, can't be corrected.

The possible way out seems to use the Expert mode where all local settings for conversion are disabled. I think the probable command-line in Expert mode are FFMPEG commands — which I don't now how to use.

Not saying that you try it, but sharing the information with you and @pandy and of course the visitors of this forum.


P.S. I still feel that locating the Out folder on physically another hard disk will speed things up.

[pandy]

But, the sad part is: the defects in audio, as you pointed out in earlier posts, can't be corrected.

The possible way out seems to use the Expert mode where all local settings for conversion are disabled. I think the probable command-line in Expert mode are FFMPEG commands — which I don't now how to use.

Not saying that you try it, but sharing the information with you and @pandy and of course the visitors of this forum.


P.S. I still feel that locating the Out folder on physically another hard disk will speed things up.

You may try to use different decoder in ffmpeg - some of them have different error resilience/concealment capabilities (try to force particular decoder, please specify its name before input), consider to use external mp2 decoder, finally you may play with ffmpeg error resilience/concealment (perhaps this is easiest to start) - look at:

  -err_detect        <flags>      .D.VA... set error detection flags (default 0)
     crccheck                     .D.VA... verify embedded CRCs
     bitstream                    .D.VA... detect bitstream specification deviations
     buffer                       .D.VA... detect improper bitstream length
     explode                      .D.VA... abort decoding on minor error detection
     ignore_err                   .D.VA... ignore errors
     careful                      .D.VA... consider things that violate the spec, are fast to check and have not been seen in the wild as errors
     compliant                    .D.VA... consider all spec non compliancies as errors
     aggressive                   .D.VA... consider things that a sane encoder should not do as an error

 e.g. -err_detect aggressive

And yes, using different HDD will speed-up whole process by reasonable factor (higher gain for fast CPU but for slow CPU there will be gain also).

This may work:

@ffmpeg.exe -err_detect aggressive -fflags +igndts+genpts -y -i %1 -c copy -fflags +genpts -f matroska %~n1_ec.mkv

[ConverterCrazy]

I can confirm the latest Sherpya build of ffmpeg runs fine on XP and the batch file from post #23 works (I added MKV as an input type).

When the source was the mpeg sample, ffmeg offered warnings about the queue input being backwards in time (in relation to the MP3 encoding), but the output seemed fine.

The problem with the frame rate and aspect ratio seems to be caused by remuxing the source as MKV. I only tested with MKVMergeGUI but it's not setting the correct aspect ratio automatically when remuxing and the output file is reported by MediaInfo to be 50fps even though it plays normally.
If I specify 4:3 and 25p when remuxing the mpeg sample as MKV, that MKV as the source results in a 25fps AVI with a 4:3 aspect ratio.

Personally, I'd resize to square pixels rather than set an aspect ratio as it's likely a hardware player won't obey the aspect ratio for an AVI. I tried adding the scale function to resize it but ffmpeg either gave me a cropped output or resized while setting a new aspect ratio. I guess I don't know how to use the scale function properly but I wasn't motivated to spend a lot of time on it.

Would the source files in question be exactly 4:3 or should they be ITU 4:3? I suspect it's the latter in which case I think the source aspect ratio should be 1.367 rather than 1.333

As a side note..... I downloaded the latest version of Hybrid a few days ago but it wouldn't run, spitting out an error regarding ffmpeg and msvcrt.dll. After replacing ffmpeg.exe in the Hybrid folder with the Sherpya build, Hybrid now runs.

Cheers.

@hello_hello
I'm back to share info. But, before that I need your help with audio cleaning — the process you had suggested using FFMPEG. Obviously, I am not able to run your script.

I want a command that will handle only the audio (as you had diagnosed and suggested).
In my little understanding the script should run like:

For %%1 in (*.avi) do (@ffmpeg.exe -analyzeduration 50M -i "%%1" -c copy… (audio related commands) 

I mean copy video and process audio only.
Is it possible?

Thanks.

[pandy]

I want a command that will handle only the audio (as you had diagnosed and suggested).
In my little understanding the script should run like:

For %%1 in (*.avi) do (@[url=http://www.videohelp.com/software/ffmpeg]ffmpeg[/url].exe -analyzeduration 50M -i "%%1" -c copy… (audio related commands) 

I mean copy video and process audio only.
Is it possible?

Thanks.

You may copy video and apply processing and encoding to audio.

use "-c:v" copy to copy video, you need to specify audio codec "-c:a" and audio processing "-af"

[ConverterCrazy]

I want a command that will handle only the audio (as you had diagnosed and suggested).
In my little understanding the script should run like:

For %%1 in (*.avi) do (@[url=http://www.videohelp.com/software/ffmpeg]ffmpeg[/url].exe -analyzeduration 50M -i "%%1" -c copy… (audio related commands) 

I mean copy video and process audio only.
Is it possible?

Thanks.

You may copy video and apply processing and encoding to audio.

use "-c:v" copy to copy video, you need to specify audio codec "-c:a" and audio processing "-af"

Hello @pandy and @hello_hello

Thank you so much for bearing with me for so long.

This what I have done:

1 - I changed container of all the MPG files to MKV —without recoding — using one of your FFMPEG scripts. I ignored all those incomprehensible errors.

2 - Then using AnotherGUI, I converted them to AVI — XVID for video and MP3 for audio. The audio was too loud… jarred my ear drums.

3 - I used the script (actually a BATCH file) from this thread https://forum.videohelp.com/threads/255147-Mini-Guide-Normalizing-Audio-for-Multiple-AVI-files, set audio track gain at 80dB (that is the acceptable audio volume after 2 trials at 86 and 83). This script uses command line options for MP3Gain which I got from https://sourceforge.net/p/mp3gain/discussion/164669/thread/ae108c97/ .

I don't know if this was the best approach. But, I am nearly satisfied with the results.

I have to do this with my collection of DVDs I bought at a hefty price and with ear-splitting audio. My above method will not work with DVDs — for I prefer AC3 audio and not MP3. But, that is the subject matter of another thread.

And, @pandy, my apologies that I could not try your above-cited suggestion.


I sincerely thank you both and the VideoHelp forum.

[pandy]

2 - Then using AnotherGUI, I converted them to AVI — XVID for video and MP3 for audio. The audio was too loud… jarred my ear drums.

3 - I used the script (actually a BATCH file) from this thread https://forum.videohelp.com/threads/255147-Mini-Guide-Normalizing-Audio-for-Multiple-AVI-files, set audio track gain at 80dB (that is the acceptable audio volume after 2 trials at 86 and 83). This script uses command line options for MP3Gain which I got from https://sourceforge.net/p/mp3gain/discussion/164669/thread/ae108c97/ .

Adjusting peak level bellow -6.0206dBFS is pure waste of bits - if audio is too loud adjust volume level at the amplifier - personally i care only about avoiding clipping in digital domain - perceived audio level i.e. loudness is completely different topic but i believe nowadays we are all victims of the noisy environment and loudness war.

[ConverterCrazy]

@pandy

You mean I didn't do right by normalizing volume to 80dB?

What else can I do?
Though I understand what you mean by

care only about avoiding clipping in digital domain - perceived audio level

[pandy]

@pandy

You mean I didn't do right by normalizing volume to 80dB?

What else can I do?
Though I understand what you mean by

care only about avoiding clipping in digital domain - perceived audio level

Well - IMHO - normalization shall be made in a way to fully use format capacity and avoid any problems - as such in PCM domain normalization shall be done for Peak level and always safe level is -6.0206dBFS.

Loudness normalization imply or inefficient format usage or dynamic processing (compression/expansion).
From my perspective adjusting in PCM domain for Loudness is not OK - loudness aligning should be performed at analog domain based on metadata but never in PCM domain as it will be lossy.

[ConverterCrazy]

@pandy

You mean I didn't do right by normalizing volume to 80dB?

What else can I do?
Though I understand what you mean by

care only about avoiding clipping in digital domain - perceived audio level

Well - IMHO - normalization shall be made in a way to fully use format capacity and avoid any problems - as such in PCM domain normalization shall be done for Peak level and always safe level is -6.0206dBFS.

Loudness normalization imply or inefficient format usage or dynamic processing (compression/expansion).
From my perspective adjusting in PCM domain for Loudness is not OK - loudness aligning should be performed at analog domain based on metadata but never in PCM domain as it will be lossy.

Aha!

I think that's the way you put it in digital audio language. To me, in analog/electronics jargon, it means don't use attenuation, take full sine wave output from the amplifier circuit and use volume control.

That's the way it should be.

[pandy]

Aha!

I think that's the way you put it in digital audio language. To me, in analog/electronics jargon, it means don't use attenuation, take full sine wave output from the amplifier circuit and use volume control.

That's the way it should be.

It means that any volume/level change in PCM domain can be lossy (especially for lowering level) and it should involve proper error (requantization) processing (dithering + noiseshaping) - usually this can't be considered as loss less (exception can be going from lower to higher bitdepth but there are some problems here also.

Level adjustment in analog domain is loss less. Loudness processing mean or changing level by fraction (requantization and outcome of RMS) or by applying dynamic compression/expansion (i.e. non linear amplification/attenuation) - both affected by requantization - side to this this affecting signal it self.

And generally yes, to fully use PCM, push your signal level to safe maximum.

[ConverterCrazy]

Thanks @pandy.

@hello_hello

I ran your FFMPEG script in #45 on another video (with copy option for video track) without a hitch and it corrected the audio.

The script didn't work on the other videos for which I had sought help. The likely reason I can think of is the bad original video track in those files. I'll try to understand FFMPEG commands.

Please let me know if you need more feedback.