Revision Date: October 30, 2018 (Europe: Paris), October 29, 2018 (America: Los Angeles), October 30, 2018 (US: Mountain)
Let me put this at the top. This will absolutely result in better quality then windows even with windows having its settings and tweaks maxed out. I am not sure why this occur. I think it has to do with driver programming
So we like audio. We especially like GOOD quality audio and as much as that is also dependent on the source. We would ideally like our operating system to not interfere with it unless its to resample it higher. In this guide I will be going through my pulse audio config in an effort to demonstrate the settings I have found most effective. This also depends on hardware and sound card quality/capabilities as well.
AUDIOPHILES : I know I am leaving quiet a bit of additional information out here. This is intended for the “average avid music listener” LOL
My config is at the bottom. The top one is the two channel default.
So lets begin with our pulse audio config. There are two ways to do this. You can either edit your user specific configuration (safer) or modify directly in edit to config.
In any account I would like to do so in edit to config as it applies system wide and its easy to make a backup.
sudo nano /etc/pulse/daemon.conf
The following settings we will modify are: (default)
default-sample-format = s16le
default-sample-rate = 48000
alternate-sample-rate = 44100
default-sample-channels = 2
default-channel-map = front-left,front-right
default-fragments = 4
default-fragment-size-msec = 25
resample-method = speex-float-1
enable-lfe-remixing = no
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 9
rlimit-rtprio = 9
daemonize = no
Lets talk about what those settings are before they are blindly modified.
Lets begin with the default-sample-format :
This is the default sampling format of the sampler. The quality will be different for each sample format. This is because each sampler has some slightly different mathematics. s samplers are fixed point samplers where ulaw alaw and u8 filters are older 8bit PCM samplers I would avoid using them for obvious reasons… Floats offer a very high level of precision however it is still dependent on how precise your source is. (assuming good hardware) For me, float32le appears to be producing the highest quality of sound.
When determining sample format you must select it based upon your CPU architecture’s byte order or also called Endianness. You can determine your CPU’s byte order by using below command: lscpu | grep 'Byte Order'
Here is a list of sample formats:
u8, s16le, s16be, s24le, s24be, s24-32le, s24-32be, s32le, s32be float32le, float32be, ulaw, alaw
What about default-sample-rate and alternate-sample-rate?:
This determines in Analog to Digital Converter (ADC) or Digital to Analog Converter (DAC) sampling rate and the alternative sample rate. The higher the rate of the source typically the better the quality of audio IF done correctly. So I definitely would like to set the bar high and rather upsample then downsample. The sound system will determine which to use, either the default or alternative automatically.
In my configuration, I have used 192000Hz. This is more than enough to get higher quality sound. We can speak more about sampling if enough people would like to understand some rudimentary signal processing (I WOULD LOVE TO GEEK OUT ABOUT IT)
The next setting is default-sample-channelsanddefault-channel-map:
The number of channels to be sampled and their output mapping. I use a 5.1 system so I have selected the number of channels to be 5.1 and channel map to suit it. In case if you have 2.1 or 5.1 surround sound system, you will have to consult the pulse documentation to figure out how to do tweak the configuration or see mine for 5.1. It is just a case of identifying speakers and stuff.
The next setting I am modifying is default-fragmentsanddefault-fragment-size-msec:
Some hardware drivers require the hardware playback buffer to be subdivided into several fragments. These configurations determines the number of fragments and a duration of a single fragment. Defaults are 4 and 25ms so the total buffer will be 100ms long. I have selected 2 and 125ms. If you have a good sound card you can ignore this configuration since most of newer sound drivers support timer-base scheduling which is a god send. Gone are the days of locking up your audio system because of hardware buffer issues
Moving onto the next setting we have resample-method:
The resampling algorithm to use when requiring up or down sampling math applied to the signal. I have selected and gone for broke speex-float-10. It is a point sampler, which offers better sound quality than speex-fixed* methods however it is CPU intensive. On a single core it will use upwards of 8-12 percent instead of less then 1.
Available values:
trivial speex-float-0 speex-float-1 speex-float-2 speex-float-3 speex-float-4 speex-float-5 speex-float-6 speex-float-7 speex-float-8 speex-float-9 speex-float-10 speex-fixed-0 speex-fixed-1 speex-fixed-2 speex-fixed-3 speex-fixed-4 speex-fixed-5 speex-fixed-6 speex-fixed-7 speex-fixed-8 speex-fixed-9 speex-fixed-10 ffmpeg auto copy peaks
The next setting is enable-lfe-remixing:
This determines the upmixing or downmixing channel behaviour. If disabled output LFE channel is available the signal on the input LFE channel will be ignored. This output is for moving the LFE to the subwoofer instead of deferred mixing into 2.0 and 4.0 speakers. I have a 5.1 system so I need it on. The Low-frequency effects (LFE) channel is the name of an audio track specifically intended for deep, low-pitched sounds ranging from 3–120 Hz. No it wont boost your bass on your headphones LOL.
We do need to make a small modification to the scheduling configuration of pulse daemon:
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 9
rlimit-rtprio = 9
daemonize = no
high-priority setting makes the pulse daemon is a high priority process. I would advice against using any kind of real time processing. This can cause the system to lock up and high priority should suffice. I have personally experienced it lock up a 16 core processor system so BE ADVISED. This is accomplished by increasing the realtime-priority much higher
Lets create direct communication between pulse and the kernel driver. This should reduce our latency. Run the following command:
$ sudo nano /etc/asound.conf
The default configuration will be attempting to use pulseaudio by default. Change the pulse audio section or create the config and make sure it reads as follows
# Use PulseAudio plugin hw
pcm.!default {
type plug
slave.pcm hw
}
slave.pcm hw plugin will communicates directly with ALSA kernel driver. This means you have completely unaltered communication to the kernel audio driver. This will reduce the latency involved in higher quality conversion
Then reboot your computer or restart Pulse and ALSA processes.
Here is my configuration: (LFE is sub)
default-sample-format = float32le
default-sample-rate = 19200
alternate-sample-rate = 96000
default-sample-channels = 5.1
default-channel-map = front-left, front-right, rear-left, rear-right, center, LFE
default-fragments = 2
default-fragment-size-msec = 125
resample-method = speex-float-10
enable-lfe-remixing = yes
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 9
rlimit-rtprio = 9
daemonize = no