Is there a way of leveling/compressing the sound system-wide?

Question

As a Laptop user, I'm sure that a lot of people, even the ones using Netbooks would have already gone through this problem. Especially when listening to podcasts, and using it as an example, the sound might have loud moments and quiet moments, one person speaking loud and the other speaking very quiet in the same episode. Thereby, I always wanted the sound to be compressed system-wide, and I just noted the other day, that in Windows, some Realtek drivers already offer this function.

We have already a pulseaudio plugin for equalization system-wide, and although it still has problems like not letting us change its values and listen the change in real time, or cracking the sound while changing pulse volume, I do love it and use it. Now I'm just missing something to keep the sound around 0db (or near the volume level you're using) as a leveler plugin for pulseaudio.

Any suggestion?

Answer 1

I had success with the example shown in this answer.

Install Steve Harris's LADSPA plugins and then run the following in pacmd:

load-module module-ladspa-sink sink_name=compressor plugin=sc4m_1916 label=sc4m control=1,1.5,401,-30,20,5,12
set-default-sink compressor

The parameters (the control=1,1.5,401,-30,20,5,12 part above) for this compressor are described in Steve Harris' LADSPA Plugin Docs:

1. RMS/peak: The balance between the RMS and peak envelope followers.RMS is generally better for subtle, musical compression and peak is better for heavier, fast compression and percussion.
2. Attack time (ms): The attack time in milliseconds.
3. Release time (ms): The release time in milliseconds.
4. Threshold level (dB): The point at which the compressor will start to kick in.
5. Ratio (1:n): The gain reduction ratio used when the signal level exceeds the threshold.
6. Knee radius (dB): The distance from the threshold where the knee curve starts.
7. Makeup gain (dB): Controls the gain of the makeup input signal in dB's.
8. Amplitude (dB): The level of the input signal, in decibels.
9. Gain reduction (dB): The degree of gain reduction applied to the input signal, in decibels.

Due to a limitation of PulseAudio, it is not possible to adjust them in real time.

To experiment with different parameters, I also loaded the compressor as a real-time adjustable ALSA plugin via Alsaequal by creating the following ~/.asoundrc:

ctl.compressor {
  type equal;
  library "/usr/lib/ladspa/sc4m_1916.so";
  module "sc4m";
}

pcm.plugcompressor {
  type equal;
  slave.pcm "plug:pulse";
  library "/usr/lib/ladspa/sc4m_1916.so";
  module "sc4m";
}

pcm.compressor {
  type plug;
  slave.pcm plugcompressor;
}

A sample MP3 file can be played through the compressor using mpg321 ,

mpg321 -a hw:compressor "04 - Love Song for Yoshimi.mp3"

while alsamixer -D compressor can be used to adjust parameters in real-time.

This answer explains how to load the plugin permanently.

Answer 2

These have been extremely helpful. Thank you for the inspiration. Although it is imperfect at the moment, I have a bit to contribute back. I looked in The Steve Harris Documentation to find that there are several flavors of the compressor, and I chose to use the stereo one. It did complicate using Pulse Audio Volume Control to place the compressor on multiple applications like Chrome and VLC, but I like the result. My motivation for using a compressor is to place a very strict lower and upper limit on volume. In order to not wake anyone in the house late at night, I didn't want to be constantly turning up the volume to hear the characters speaking in media, only to rush to turn it back down whenever superhero action happens or a commercial comes on. I followed the inspiration here and adjusted the values in realtime. Then I took my desired values and put them back into the sample code. In sum, the sample code I am contributing shamelessly borrows from the above example, but features the stereo compressor with a pretty strict upper and lower volume limit. Lastly, I put it into the form of a script, which is not fully working. It does not like the set_default line. I hope this is useful to the community.

#!/bin/sh
#ComperssorScript.sh
#Script to start PulseAudio Compressor with desired settings
#Original: 2016 September 17

pacmd load-module module-ladspa-sink sink_name=compressor plugin=sc4_1882 label=sc4 control=9,5,63,-6,15,3,49
set-default-sink compressor

#The parameters (the control=1,1.5,401,-30,20,5,12 for example) for this #compressor are described in Steve Harris' LADSPA Plugin Docs:
#RMS/peak: The balance between the RMS and peak envelope followers. RMS #is generally better for subtle, musical compression and peak is better #for heavier, fast compression and #percussion.
#9,
#Attack time (ms): The attack time in milliseconds.
#5,
#Release time (ms): The release time in milliseconds.
#63,
#Threshold level (dB): The point at which the compressor will start to #kick in.
#6,
#Ratio (1:n): The gain reduction ratio used when the signal level exceeds #the threshold.
#-15,
#Knee radius (dB): The distance from the threshold where the knee curve #starts.
#3,
#Makeup gain (dB): Controls the gain of the makeup input signal in dB's.
#49,
#Amplitude (dB): The level of the input signal, in decibels.
#no value was placed here
#Gain reduction (dB): The degree of gain reduction applied to the input #signal, in decibels.
#Due to a limitation of PulseAudio, it is not possible to adjust them in #real time.
#no value was placed here

Answer 3

Here is the stereo plugin with better control settings (Much less clipping) following ændrük's answer that uses the mono plugin example.

load-module module-ladspa-sink sink_name=compressor plugin=sc4_1882 label=sc4 control=1,1.5,300,-20,3,1,10