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Clamarc
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I applied a compressor in Music1 with the parameters:

Threshold: -25.00 db
Ratio  : 2.00:1
Attack : 0.25 ms
Release: 20.0 ms 

The result of the volumedetect function was:

Output #0, null, to 'NUL':
  Metadata:
    title           : Music1
    TKEY            : F#m
    comment         : 
    album           : Electro's Remix
    genre           : Electro
    artist          : Chic
    encoder         : Lavf59.17.102
  Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
      Metadata:
        encoder         : Lavc61.5.104 pcm_s16le
[Parsed_volumedetect_0 @ 0000021002da5180] n_samples: 30392156
[Parsed_volumedetect_0 @ 0000021002da5180] mean_volume: -17.1 dB
[Parsed_volumedetect_0 @ 0000021002da5180] max_volume: -0.9 dB
[Parsed_volumedetect_0 @ 0000021002da5180] histogram_0db: 2
[Parsed_volumedetect_0 @ 0000021002da5180] histogram_1db: 9

Note above that the max_volume value has changed to -0.9 db

I performed audio normalization with the parameter LUFS = -10.0 and the True Peak = -0.0:

Input Integrated:    -14.6 LUFS
Input True Peak:      -0.7 dBTP
Input LRA:             4.6 LU
Input Threshold:     -24.6 LUFS

Output Integrated:   -10.5 LUFS
Output True Peak:     +0.0 dBTP
Output LRA:            3.8 LU
Output Threshold:    -20.5 LUFS

Normalization Type:   Dynamic
Target Offset:        +0.5 LU

Note above that the result of Output Integration was much better.

I applied a compressor in Music1 with the parameters:

Threshold: -25.00 db
Ratio  : 2.00:1
Attack : 0.25 ms
Release: 20.0 ms 

The result of the volumedetect function was:

Output #0, null, to 'NUL':
  Metadata:
    title           : Music1
    TKEY            : F#m
    comment         : 
    album           : Electro's Remix
    genre           : Electro
    artist          : Chic
    encoder         : Lavf59.17.102
  Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
      Metadata:
        encoder         : Lavc61.5.104 pcm_s16le
[Parsed_volumedetect_0 @ 0000021002da5180] n_samples: 30392156
[Parsed_volumedetect_0 @ 0000021002da5180] mean_volume: -17.1 dB
[Parsed_volumedetect_0 @ 0000021002da5180] max_volume: -0.9 dB
[Parsed_volumedetect_0 @ 0000021002da5180] histogram_0db: 2
[Parsed_volumedetect_0 @ 0000021002da5180] histogram_1db: 9

Note above that the max_volume value has changed to -0.9 db

I performed audio normalization with the parameter LUFS = -10.0 and the True Peak = -0.0:

Input Integrated:    -14.6 LUFS
Input True Peak:      -0.7 dBTP
Input LRA:             4.6 LU
Input Threshold:     -24.6 LUFS

Output Integrated:   -10.5 LUFS
Output True Peak:     +0.0 dBTP
Output LRA:            3.8 LU
Output Threshold:    -20.5 LUFS

Normalization Type:   Dynamic
Target Offset:        +0.5 LU

Note above that the result of Output Integration was much better.

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Clamarc
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I analyzed the Music1 audio using the volumedetect function, see the results below:

Output #0, null, to 'NUL':
  Metadata:
    title           : Music1
    TKEY            : F#m
    comment         : 
    album           : Electro's Remix
    genre           : Electro
    artist          : Chic
    encoder         : Lavf59.17.102
  Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc59.21.100 pcm_s16le
size=N/A time=00:00:00.02 bitrate=N/A speed=2.61e+04x    
size=N/A time=00:04:30.26 bitrate=N/A speed= 541x    
size=N/A time=00:05:44.60 bitrate=N/A speed= 543x    
video:0kB audio:59364kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[Parsed_volumedetect_0 @ 0000022928c2e1c0] n_samples: 30394368
[Parsed_volumedetect_0 @ 0000022928c2e1c0] mean_volume: -15.7 dB
[Parsed_volumedetect_0 @ 0000022928c2e1c0] max_volume: 0.0 dB
[Parsed_volumedetect_0 @ 0000022928c2e1c0] histogram_0db: 13190
[Parsed_volumedetect_0 @ 0000022928c2e1c0] histogram_1db: 37702

In the result above that the max_volume parameter has a value of 0.0 dB.
Could this be the problem with normalization not reaching -10.0 that I set?

I analyzed the Music1 audio using the volumedetect function, see the results below:

Output #0, null, to 'NUL':
  Metadata:
    title           : Music1
    TKEY            : F#m
    comment         : 
    album           : Electro's Remix
    genre           : Electro
    artist          : Chic
    encoder         : Lavf59.17.102
  Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc59.21.100 pcm_s16le
size=N/A time=00:00:00.02 bitrate=N/A speed=2.61e+04x    
size=N/A time=00:04:30.26 bitrate=N/A speed= 541x    
size=N/A time=00:05:44.60 bitrate=N/A speed= 543x    
video:0kB audio:59364kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[Parsed_volumedetect_0 @ 0000022928c2e1c0] n_samples: 30394368
[Parsed_volumedetect_0 @ 0000022928c2e1c0] mean_volume: -15.7 dB
[Parsed_volumedetect_0 @ 0000022928c2e1c0] max_volume: 0.0 dB
[Parsed_volumedetect_0 @ 0000022928c2e1c0] histogram_0db: 13190
[Parsed_volumedetect_0 @ 0000022928c2e1c0] histogram_1db: 37702

In the result above that the max_volume parameter has a value of 0.0 dB.
Could this be the problem with normalization not reaching -10.0 that I set?

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Clamarc
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Ffmpeg LUFS Loudnorm normalization results

I use the Loundnorm filter in Ffmpeg to normalize LUFS in a folder containing three audios, Music1.mp3, Music2.mp3 and Music3.mp3 using the double-pass method.

The LUFS and True Peak values are the same for all audio files:
LUFS -10.0 and True Peak -0.5

Code:

md "C:\Users\%username%\Desktop\Temp_normalizing_lufs"    ------> temp folder audio files being normalized   
pushd "%Userprofile%\Desktop\Audios LUFS"                 ------> folder with the original audio files

                                                                                                      __
FOR /F "delims=" %%a in ('where .:*.mp3 ^|findstr /vi "_LOUDNORM  _EBU"') DO (                          |                
  SET "filename=%%~na"                                                                                  |
  ffmpeg -hide_banner -i "%%a" -af "[0:a]loudnorm=print_format=summary" -f null NUL 2> "%%~na.log"      |   
  @FOR /F "tokens=3" %%b IN ('FINDSTR /C:"Input Integrated" "%%~na.log"') DO (SET II=%%b)               |
  @FOR /F "tokens=4" %%b IN ('FINDSTR /C:"Input True Peak" "%%~na.log"') DO (SET ITP=%%b)               |  set original audio files values  
  @FOR /F "tokens=3" %%b IN ('FINDSTR /C:"Input LRA" "%%~na.log"') DO (SET ILRA=%%b)                    |  to use as parameters in loudnorm
  @FOR /F "tokens=3" %%b IN ('FINDSTR /C:"Input Threshold" "%%~na.log"') DO (SET IT=%%b)                |
  @FOR /F "tokens=3" %%b IN ('FINDSTR /C:"Target Offset" "%%~na.log"') DO (SET TO=%%b)                  |
  DEL "%%~na.log"                                                                                     __|

  SETLOCAL ENABLEDELAYEDEXPANSION
  FOR /F "tokens=1,2 delims=," %%b IN ('ffprobe -v 0 -select_streams a -show_entries "stream=bit_rate,sample_rate" -of "csv=p=0" "!filename!.mp3"')     ----> getting the sample rate and bitrate of the original audio file to use as parameters in loudnorm
  DO (  
      ffmpeg -hide_banner -i "!filename!.mp3" -af "loudnorm=linear=true:I=!-10.0!:LRA=11:tp=!-0.5!:measured_I=!II!:measured_LRA=!ILRA!:measured_tp=!ITP!:measured_thresh=!IT!:offset=!TO!:print_format=summary" -c:v copy -id3v2_version 3 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (front)" -acodec mp3 -b:a %%c -ar:a %%b "C:\Users\%username%\Desktop\Temp_normalizing_lufs\!filename!.mp3"
     )
  ENDLOCAL
)

xcopy "C:\Users\%username%\Desktop\Temp_normalizing_lufs\*.mp3" "C:\Users\%username%\Desktop\Normalized Audios Lufs\LOUDNORM\MP3\LUFS %-10.0%" /y /s /i   ----> copying the audio files from the temporary folder to the final folder

                                                                __
del /q "C:\Users\%username%\Desktop\Temp_normalizing_lufs\*.*"    |
rmdir "C:\Users\%username%\Desktop\Temp_normalizing_lufs" /s /q   |  Removing the temporary folder from the desktop
                                                                __|

Result of the Music1 audio file:

Input Integrated:    -13.3 LUFS
Input True Peak:      +1.6 dBTP
Input LRA:             6.1 LU
Input Threshold:     -23.4 LUFS

Output Integrated:   -11.8 LUFS
Output True Peak:     -0.5 dBTP
Output LRA:            4.5 LU
Output Threshold:    -21.9 LUFS

Normalization Type:   Dynamic
Target Offset:        +1.8 LU

Why doesn't the above value in the Output Integrated reach the LUFS of -10.0?
PS: even if I change the LUFS to -9.0, the Intregede Output value remains the same -11.8 LUFS.

Results of the Music2 audio file:

Input Integrated:     -6.0 LUFS
Input True Peak:      +4.9 dBTP
Input LRA:             4.8 LU
Input Threshold:     -16.3 LUFS

Output Integrated:   -10.0 LUFS
Output True Peak:     -0.5 dBTP
Output LRA:            4.3 LU
Output Threshold:    -20.3 LUFS

Normalization Type:   Dynamic
Target Offset:        +0.0 LU

Results of the Music3 audio file:

Input Integrated:     -5.9 LUFS
Input True Peak:      +0.6 dBTP
Input LRA:             4.5 LU
Input Threshold:     -16.0 LUFS

Output Integrated:   -10.0 LUFS
Output True Peak:     -3.5 dBTP
Output LRA:            4.5 LU
Output Threshold:    -20.1 LUFS

Normalization Type:   Linear
Target Offset:        -0.0 LU

Why in the above audio files, Music2 uses Normalization Type: Dynamic and Music3 uses Normalization Type: Linear?... what rule defines when it is Dynamic and when it is Linear?