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Topic: R128Norm (Read 58296 times) previous topic - next topic
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R128Norm

Reply #50 –
You do not need to disable ReplayGain to use this normalizer. From your post I kind of got the impression you consider this a ReplayGain replacement. If you merely want to see how R128 loudness scanner's result differ from the old ReplayGain scanner just scan your files again with the built-in scanner. It uses the R128 method nowadays.

R128Norm

Reply #51 –
Is this possible to make any configuration for this component available? Maybe it will help tune this plugin to work better? It could be nice to listening music at night with this plugin , but I noticed that mostly it works bit to slow. When there is quieter part in track, at first it is played at low volume, after some time (about 1 - 1,5s) quiet becomes loud enough. When loud parts come back, then at first moment they are bit louder than they should be, but after some time (like several seconds) they have proper volume. Other thing is that I have impression that quiet parts, after making them louder, are even louder than parts that are regularly loud. Is there any update for this plugin expected?

R128Norm

Reply #52 –
Quote from: EpicForever on
Is this possible to make any configuration for this component available? Maybe it will help tune this plugin to work better? It could be nice to listening music at night with this plugin , but I noticed that mostly it works bit to slow. When there is quieter part in track, at first it is played at low volume, after some time (about 1 - 1,5s) quiet becomes loud enough. When loud parts come back, then at first moment they are bit louder than they should be, but after some time (like several seconds) they have proper volume. Other thing is that I have impression that quiet parts, after making them louder, are even louder than parts that are regularly loud. Is there any update for this plugin expected?


For symphonic, classical, ambiant, etc, I still use V-Level. I think kode54 said somthing about being able to change "some" settings in the next release.

R128Norm

Reply #53 –
For symphonic you've said... How about for progressive and melodic trance?

R128Norm

Reply #54 –
I can't edit my older post so I have to double post. For trance vlevel also works better... On default settings. Even on Sunny Lax'es "M.I.R.A." which has serious level dropdown in the middle of the track vlevel works very good, as well as on several Oliver Smith tracks. Or on Kyau vs Albert's "Falling Anywhere". On Original Mix of Menno de Jong's "Guanxi", Nitrous Oxide's "Morning Light"... Vlevel everywhere works better for me.

R128Norm

Reply #55 –
Really like this component, it's almost perfect. Using it when converting different songs for one use case. MP3 player for sport, USB flash drive for in the car, etc.

My car stereo lacks some performance unfortunately. The default reference loudness level makes sense, but I'd like to be able to adjust it for in this case. The files are all so low after conversion, that my weak sauce of what can barely be called a car stereo won't produce an acceptable level of loudness for my needs.

I know I can boost the signal with otherwise, but I'm not into the whole matter and would prefer to keep it as simple as possible.

tl;dl: Suggestion - Include an "oomph mode" that produces much higher output levels by choosing a higher reference loudness level.

Thanks for all your time and work you put into all these countless tools. Much appreciated.

R128Norm

Reply #56 –
I use VST plugin to boost sound..
But possibility to choose a higher loudness level would be great..

R128Norm

Reply #57 –
Sure, adding more DSPs/VSTs are an alternative do increase volume/loudness. But since I'm really happy with how the R128 stuff does its thing, there's actually no need for additional DSPs, if you could just select different reference levels as a setting.

R128Norm

Reply #58 –
Here's what I'd do (until r128Norm has an option to alter reference levels). This DSP chain can be setup for conversion only:

DSPs:
  • r128Norm
  • Graphic Equalizer  (just to boost volume) say +6dB
  • Advanced Limiter (this will ensure no samples clip)

There may well be better ways to do this - but I do something similar and this works for me.

C.
PC = TAK + LossyWAV  ::  Portable = Opus (130)

R128Norm

Reply #59 –
Sounds like a reasonable approach. Thanks for the suggestion.

Re: R128Norm

Reply #60 –
I started using the foo_r128norm component in foobar about 3 weeks ago. I like how it handles flac, ape, wv files etc. but when I play an SACD iso file it usually comes out too quiet. Perhaps kode54 could test this for me?

Re: R128Norm

Reply #61 –
If you're using the SACD component's ASIO output method, it bypasses all DSPs and goes straight to your sound device. Or at least I assume as much, since DSP chains rightly can't process DSD content directly.

Re: R128Norm

Reply #62 –
@kode54 is there a chance you make standalone app for this dsp ?
qaac -cvbr 0 -he

Re: R128Norm

Reply #64 –
I don't even see how that would work. Maybe it would be more constructive to ask the author of VB-Audio Voicemeeter Banana to add such a thing to their software.

Re: R128Norm

Reply #65 –
just curious, does this compressor would be louder if audio source is mono than stereo?

reproduce:
insert 'downmix channels to mono' before this compressor.

Re: R128Norm

Reply #66 –
Thank you.
Nice plugin like Vlevel, but more advanced.
The question is, your plugin do any change to dynamic range of track?
And question num 2, your plugin do any compression to sound, when is playing?

Re: R128Norm

Reply #67 –
It may affect the dynamic range, since it tries to keep a near constant volume level, so quiet parts will become louder, and louder parts may become quieter. It does not do any equalization type compression, however, nor does it reshape the peaks to squash them, it should lower the volume overall instead.

Re: R128Norm

Reply #68 –
While waiting for the arrival of new headphones for portable audio player, I have to use broken ones to listen for a course of Youtube-downloaded lectures. Broken headphones are barely audible, yet amping the volume of sound files does the trick more or less. Question is — in this particular case when irreversible changes are acceptable — whether it is better a) to transcode MP4 into MP3 and apply Replaygain to file content or b) to transcode MP4 into MP3 with EBU R128 in DSP?
• Join our efforts to make Helix MP3 encoder great again
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• Let's pray that D. Bryant improve WavPack hybrid, C. Helmrich update FSLAC, M. van Beurden teach FLAC to handle non-audio data

Re: R128Norm

Reply #69 –
For those who like to dig under the hood, below are the five minute cuts of a two hours lecture mentioned in the previous message.
Cuts are made according to the following pattern: FFmpeg -i Lecture.Original.Full.mp3 -ss 00:00:29 -t 296 -codec copy Lecture.Original.Cut.mp3
MP3Gain cut is omitted, because, to my astonishment, it did not impact that period (even its checksum matches Original cut) and in general had a subtle impact.
Waveforms are generated via nameless online tool, resized via IrfanView and optimized via Pingo.


Lecture.Original.mp3 (transcoded from AAC into MP3 via Foobar2000 using Lame 3.100 --abr 128)


Lecture.Replaygain.mp3 (track replaygain +5.30 dB applied to Original after transcoding from AAC into MP3)


Lecture.EBUR128.mp3 (EBU R128 and Advanced Limiter used during transcoding from AAC into MP3)
• Join our efforts to make Helix MP3 encoder great again
• Opus complexity & qAAC dependence on Apple is an aberration from Vorbis & Musepack breakthroughs
• Let's pray that D. Bryant improve WavPack hybrid, C. Helmrich update FSLAC, M. van Beurden teach FLAC to handle non-audio data

Re: R128Norm

Reply #70 –
Aha, MP3Gain did not amp the volume due to “Don’t clip when doing track gain” checkbox or, if you prefer command-line interface, due to /k switch (which stands for “automatically lower Track/Album gain to not clip audio”). Similarly, if I turn on “Prevent clipping” in Foobar2000’s Replaygain settings, nothing happens and the console says “Reducing applied gain due to clipping: +5.30 dB to +0.00 dB”. Dream of increasing loudness without touching the overloaded parts is still a dream.

  

When “Prevent clipping” is off, Foobar2000 does apply the track gain, but too many annoying peaks emerge closer to the end of a lecture (not present in the five minute cut above). On the contrary, the result of EBU R128 DSP paired with Advanced Compressor is almost free of clipped samples. Audition’s analysis as follows.

  

Thus, answering my own question whether it is better — in this case — to apply Replaygain or transcode with EBU R128 in DSP, it seems the latter option produces fewer annoyances, although it is not entirely natural to hear applause from the gallery at the same volume as the speaker’s voice. Do you agree?
• Join our efforts to make Helix MP3 encoder great again
• Opus complexity & qAAC dependence on Apple is an aberration from Vorbis & Musepack breakthroughs
• Let's pray that D. Bryant improve WavPack hybrid, C. Helmrich update FSLAC, M. van Beurden teach FLAC to handle non-audio data

Re: R128Norm

Reply #71 –
If you find the R128 Compressor DSP compresses too much (it works very well, but compresses too much for me) you can try the Dynamic Audio Normaliser. It's built into ffmpeg so if you use ffmpeg as the encoder, try this as the command line in fb2k. It's what I mostly use myself:

-i - -ignore_length true -af dynaudnorm=f=150 -c:a pcm_s16le %d

This should compress a bit harder:

-i - -ignore_length true -af dynaudnorm=f=75:g=11 -c:a pcm_s16le %d

And if you don't want to output a wave file or encode with ffmpeg, this is an example command line for compressing and piping the output to QAAC (you have to set cmd.exe as the encoder). Works on XP. I'm not sure about other Windows flavours.

 /d /c c:\progra~1\foobar2000\encoders\ffmpeg.exe -i - -ignore_length true -c:a pcm_f32le -af dynaudnorm=f=150 -f wav - | c:\progra~1\foobar2000\encoders\QAAC\qaac.exe --ignorelength -s --no-optimize --no-delay -V 91 -o %d -

PS If you don't wan to compress as such, there's always MP3DirectCut. For MP3s it can be used to manually reduce the volume of any unusually load peaks. It doesn't re-encode.

Re: R128Norm

Reply #72 –
And two more compressors are available for foobar2000 - foo_dsp_dynamics and foo_dsp_vlevel

Re: R128Norm

Reply #73 –
yetanotherid, how nice to get valuable feedback from the other side of the world! Dynamic Audio Normalizer served me well and relatively fast (for example, compared to another FFmpeg’s filter called loudnorm). However, after several hours of experiments I slightly modified its execution to better meet the need of increasing loudness of lectures recorded on the stage:

/d /c ffmpeg.exe -i - -ignore_length true -af "dynaudnorm=s=30:r=0.1:p=0.9" -c:a pcm_f32le -f wav - | lame.exe --ignorelength --noreplaygain --abr 128 - %d

Subtle compression (s) is added to soften towering noises from clothes and possible transcoding glitches. RMS processing (r) is enabled to amplify voice fluctuations more harmoniously by taking into account not only the peaks but also the quieter parts. Normalization threshold (p) is lowered from default 0.95 to 0.90 to leave more headroom (around -1 dB) for future enhancing (e.g. portable player’s EQ). Frame length (f) and Gaussian window size (g) are not adjusted as you suggested, because the speaker’s breath becomes accentuated. Having said that, I welcome amendments and additions.

Rollin, thanks for mentioning other Foobar2000 compatible solutions. Alas, both dsp_dynamics and dsp_vlevel require prior knowledge of compressor’s anatomy. Lack of graphical preview or at least presets for common scenarios makes applying them correctly even harder for amateurs like myself.

  
• Join our efforts to make Helix MP3 encoder great again
• Opus complexity & qAAC dependence on Apple is an aberration from Vorbis & Musepack breakthroughs
• Let's pray that D. Bryant improve WavPack hybrid, C. Helmrich update FSLAC, M. van Beurden teach FLAC to handle non-audio data

Re: R128Norm

Reply #74 –
Ah, Hydrogenaudio allows to edit own messages for so short period of time.

Edit: r=0.1 -> r=0.12


Lecture.Dynaudnorm.mp3
• Join our efforts to make Helix MP3 encoder great again
• Opus complexity & qAAC dependence on Apple is an aberration from Vorbis & Musepack breakthroughs
• Let's pray that D. Bryant improve WavPack hybrid, C. Helmrich update FSLAC, M. van Beurden teach FLAC to handle non-audio data