Hi, I have a bunch of mp3's at various levels and want to make the 'same'.
However I don't want to a program to 'normalise' or change them in any way... just get
an idea what their level is and then manually + or - in Mp3pro Trim as needed.
Is there anything that might do this?
Otherwise I can batch load 20 or so at a time into Audition, but it'd be tedious as there's several hundred.
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Since audio files tend to vary throughout, you can use ffmpeg to get a relative indication of how many frames will be affected by the +/- setting.
The mean volume below is -18.5 but has roughly 25000 out of 11823104 total that may be distorted if raised too high.
"ffmpeg" -i "C:\video.mp4" -af volumedetect -f null NULData: [Parsed_volumedetect_0 @ 00274240] n_samples: 11823104
Data: [Parsed_volumedetect_0 @ 00274240] mean_volume: -18.5 dB
Data: [Parsed_volumedetect_0 @ 00274240] max_volume: 0.0 dB
Data: [Parsed_volumedetect_0 @ 00274240] histogram_0db: 698
Data: [Parsed_volumedetect_0 @ 00274240] histogram_1db: 2774
Data: [Parsed_volumedetect_0 @ 00274240] histogram_2db: 7673
Data: [Parsed_volumedetect_0 @ 00274240] histogram_3db: 15107
You can run an R128 scan with ffmpeg.
ffmpeg.exe -nostats -i audio.mp3 -filter_complex ebur128=peak=true -f null -
Someone will probably be able to tell you how to create a batch file to make it more automated. I'm not clever at that sort of thing.
ffplay can show you a graph, but I think it has to play through the files in real time.
ffplay.exe" -f lavfi -i "amovie=audio.mp3,ebur128=video=1:meter=18 [out0][out1]"
Or you can scan multiple files simultaneously with foobar2000. It can also adjust the volume losslessly (the only limitation is it has to be done in 1.5dB increments).
Screenshot 1 is an ffmpeg scan. The volume is -16.9 LUFS (effectively -16.9 dB).
Screenshot 2 is after a foobar2000 scan. The scan resulted in the same volume, but it's shown as the adjustment required to achieve the ReplayGain target volume of 89dB, or -18 LUFS (effectively -18 dB).
Last edited by hello_hello; 22nd Oct 2018 at 18:46.
You can try this:
for %%A in (%*) do ffmpeg -vn -nostats -filter_complex ebur128=peak=true -report -v 0 -f null - -i %%A
Drag and drop any audio file over it, to get a report like this at the source folder for each file.
ffmpeg started on 2018-10-27 at 04:16:58 Report written to "ffmpeg-20181027-041658.log" Command line: "ffmpeg" -vn -nostats -filter_complex "ebur128=peak=true" -report -v 0 -f null - -i "01. All That She Wants.m4a" ffmpeg version N-92266-gbf324359be Copyright (c) 2000-2018 the FFmpeg developers built with gcc 8.2.1 (GCC) 20181017 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth libavutil 56. 20.100 / 56. 20.100 libavcodec 58. 33.102 / 58. 33.102 libavformat 58. 19.102 / 58. 19.102 libavdevice 58. 4.106 / 58. 4.106 libavfilter 7. 37.100 / 7. 37.100 libswscale 5. 2.100 / 5. 2.100 libswresample 3. 2.100 / 3. 2.100 libpostproc 55. 2.100 / 55. 2.100 Splitting the commandline. Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'. Reading option '-nostats' ... matched as option 'stats' (print progress report during encoding) with argument 0. Reading option '-filter_complex' ... matched as option 'filter_complex' (create a complex filtergraph) with argument 'ebur128=peak=true'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '-v' ... matched as option 'v' (set logging level) with argument '0'. Reading option '-f' ... matched as option 'f' (force format) with argument 'null'. Reading option '-' ... matched as output url. Reading option '-i' ... matched as input url with argument '01. All That She Wants.m4a'. Finished splitting the commandline. Parsing a group of options: global . Applying option nostats (print progress report during encoding) with argument 0. Applying option filter_complex (create a complex filtergraph) with argument ebur128=peak=true. Applying option report (generate a report) with argument 1. Applying option v (set logging level) with argument 0. Successfully parsed a group of options. Parsing a group of options: input url 01. All That She Wants.m4a. Successfully parsed a group of options. Opening an input file: 01. All That She Wants.m4a. [NULL @ 000001bdf8b1b300] Opening '01. All That She Wants.m4a' for reading [file @ 000001bdf8b1c340] Setting default whitelist 'file,crypto' [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] ISO: File Type Major Brand: M4A [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] Unknown dref type 0x206c7275 size 12 [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] stream 0, timescale not set [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] Before avformat_find_stream_info() pos: 2056089 bytes read:2086965 seeks:0 nb_streams:2 [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] demuxer injecting skip 2112 / discard 0 [aac @ 000001bdf8b2d480] skip 2112 / discard 0 samples due to side data [aac @ 000001bdf8b2d480] skip whole frame, skip left: 1088 [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 000001bdf8b1b300] After avformat_find_stream_info() pos: 2056095 bytes read:2086965 seeks:0 frames:2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '01. All That She Wants.m4a': Metadata: major_brand : M4A minor_version : 0 compatible_brands: M4A mp42isom creation_time : 2018-06-29T14:34:38.000000Z encoder : qaac 2.65, CoreAudioToolbox 184.108.40.206, AAC-LC Encoder, TVBR q118, Quality 96 Encoding Params : vers iTunSMPB : 00000000 00000840 000001B0 0000000000908610 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 AYGAIN_REFERENCE_LOUDNESS: 89.0 dB AYGAIN_TRACK_GAIN: +0.28 dB AYGAIN_TRACK_PEAK: 0.60839844 AYGAIN_ALBUM_GAIN: +0.28 dB AYGAIN_ALBUM_PEAK: 0.60839844 DISCCONFIDENCE : 37/42;42/47 TRACKCONFIDENCE : 41/42;46/47 C : C+BB+3FA7+838B+C8F0+10150+142DF+18321+1C7B9+20C03+2572B+291C6+2DB79+3261A;C+BB+3FA7+838B+C8F0+10150+142DF+18321+1C7B9+20C03+2572B+291C6+2DB79+3261A ASE DATE : 1993-11-23 ASECOUNTRY : JP ISHER : Arista LNO : 07822-18740-2 track : 1/12 title : All That She Wants album : The Sign artist : Ace of Base disc : 1/1 date : 1992 cBrainz Album Release Date: 1993-11-23 cBrainz Album Release Country: JP L : Arista LOGNUMBER : 07822-18740-2 Duration: 00:03:34.83, start: 0.047889, bitrate: 285 kb/s Stream #0:0(und), 1, 1/44100: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 208 kb/s (default) Metadata: creation_time : 2018-06-29T14:34:38.000000Z Stream #0:1, 1, 1/90000: Video: png, rgb24(pc), 1200x1200 [SAR 2835:2835 DAR 1:1], 90k tbr, 90k tbn, 90k tbc Successfully opened the file. ... [Parsed_ebur128_0 @ 000001bdf8b7a640] Summary: Integrated loudness: I: -18.1 LUFS Threshold: -28.3 LUFS Loudness range: LRA: 3.9 LU Threshold: -38.3 LUFS LRA low: -20.4 LUFS LRA high: -16.6 LUFS True peak: Peak: -4.3 dBFS [AVIOContext @ 000001bdf8b24600] Statistics: 7654449 bytes read, 0 seeks
Last edited by amaipaipai; 27th Oct 2018 at 17:55.
Thanks very much for the replies and ideas. This is all a bit new, both running these .bat files and interpreting what they show.
But if I can get some idea of levels I can focus on just those overly high (or low).
I'm not having any luck with something.bat. I drag an mp3 file to it in explorer? That flashes the dos window momentarily.
I did copy ffmpeg.exe and dll to the same folder, but no change.
Then tried "ffmpeg.exe -nostats -i audio.mp3 -filter_complex ebur128=peak=true -f null -" changing audio.mp3 to my filename and that shows
H:\1980\A-Sides>ffmpeg.exe -nostats -i The Whispers - My Gir
l.mp3 -filter_complex ebur128=peak=true -f null -
FFmpeg version SVN-r26071, Copyright (c) 2000-2010 the FFmpeg developers
built on Dec 22 2010 04:07:12 with gcc 4.4.2
configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb
is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg
--enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb
--enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect -
-enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss
l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads --
cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack
libavutil 50.35. 0 / 50.35. 0
libavcore 0.16. 0 / 0.16. 0
libavcodec 52.100. 0 / 52.100. 0
libavformat 52.88. 0 / 52.88. 0
libavdevice 52. 2. 2 / 52. 2. 2
libavfilter 1.69. 0 / 1.69. 0
libswscale 0.12. 0 / 0.12. 0
ffmpeg.exe: unrecognized option 'nostats'
So looks better, but maybe it's an older version and should be upgraded? Then I might see the same screen as hello_helloo?
I also don't know what those numbers mean. Maybe just compare the LUFS value of each (when I see them)?
Hmmm kind of lost at that site, sorry. Could you tell me where/what to download?
BeyonWiz T3 PVR ~ Popcorn A-500 ~ Samsung ES8000 65" LED TV ~ Windows 7 64bit ~ Yamaha RX-A1070 ~ QnapTS851-4G
Thank you, got it.
There's a folder called ffmpeg-4.0.2-win32-static and 3 subfolders. Where should that go ? Does it matter ? I see no setup file.. .
My older ffmpeg is in C:\Program Files\VideoPerformer. Should I get rid of that (I don't use it)?
Do I need the DLL ?
LOL sorry for all the questions.
There's no setup file. On the VideoHelp page, "Download ffmpeg 4.0.2 Windows" is the 32 bit version. "Download ffmpeg 4.0.2 Windows 64-bit" is the 64 bit version.
Download the appropriate one, unzip the zip file, and you can just copy ffmpeg.exe from the "bin" folder to wherever you want it.
It doesn't matter, all you need to do is to extract and use the ffmpeg.exe, nothing else.
Oh, I'm using Windows XP. Would that be why ?
ffmpeg is no longer XP compatible. There's a link in this post for a recent version that is.
Why do these people make it so hard to find download links... apart from a freescan from driver.com ?
Again I can't find the program. Just pages full of rocket science talk with everything except what you want.
My last message never showed up here ! I've changed computers and now using Win 10 and teh file I got yesterday. I can get an output similar to hello hello but nothing from drag and drop like amaipaipai said.
However I can't pipe the ouput to a file e.g.
ffmpeg.exe -nostats -i u80_258a.mp3 -filter_complex ebur128=peak=true -f null - 2>&1>outputfile9.txt
writes an empty file.
At one stage it wrote a log, but I can't get that again. Is there an instruction to always wrote a log ?
If I can export the data somehow my aim now is to run a batch and compare Summarys, This may identify when files need levels adjusted.
PS Is it possible to get a Summary only ?
Make sure you copy and paste the command exactly. You should get .log files with all the info.
Download foobar2000. Load the MP3s into a playlist. Highlight them all, right click and select "ReplayGain/Scan per file Track Gain". The ReplayGain scanner uses the newer EBU R128 scanning method.
When it's done, save the results (it's saved to MP3 tags). Right click again and select "ReplayGain/Apply Track Gain to file content". You might find that's job done,
I'm pretty sure that by default, after the ReplayGain info is saved, fb2k adjusts the playback volume according to the ReplayGain info, so you'll be able to tell if the result is what you're hoping for just by playing the MP3s after saving the tags, before you losslessly adjust them. The right click menu has an option to remove the ReplayGain data from the MP3s.
The volume is adjusted losslessly, but of course you should make a backup copy of the MP3s first in case you want to go back to their original state. Foobar2000 can be installed as a "portable" version as well as a regular program.
Hi. I will certainly try that out. Need to check what's done to the tag though.
At the moment I've had some success - the ffmpeg output is piped to a text file and I grab that in Excel and write the summary to a worksheet. It's run as a batch so I can compare several at once. Looking at the files in Abode Audition there's correlation between the waveshape and the True Peak value in the ffmpeg Summary.
However your foobar suggestion may be simpler and quicker.
Just as a matter of interest though, can I use ffmpeg to increase the mp3 volume by, say, 3 db?
It certainly has worked. Attached image shows 8 files- first 4 before FB, and next 8 afterwards. It appears to reduce the loudest to match the rest. Now all could do with a slight increase.
I need to do more tests with low/high extremes levels and see what happens
There is a nice write up for Foobar and MP3tag (both freeware) use to modify the ID3 tags and explains what the tags names actually used are. It's a good page to read if you have spare time and has some good information.
Thanks Budman that was a good read, even though it's target is iTunes, it gave me some ideas. It may be easier than my VBA script with ffmpeg.
I increased vol by 9db in mp3ProTrim, and reduced it by 9db in a second mp3, then applied Foobars Track Replay Gain. Which If I understand it correctly, doesn't change anything but writes data in the tag area that's used by the player to adjust levels. (Existing tags are preserved.).
FB very accurately reversed my level changes.. (see below, before and after) but seems to set the max level to somewhat less than optimum. A 3db increase would be a good thing. From Googling ffmpeg can adjust levels but I can't find the correct command. Does anyone know it. Ideally the track gain figures from FB could be used to set this ? My goal is to have the mp3 level changed, so no input from the player is needed.