Audio in Linux becomes annoying again (continued)

Well, it turned out that the problem playing some system sounds in Thunderbird, described in my previous post, was due to PulseAudio.

Despite sounding fine when played by SMPlayer, the audio clips that sounded distorted/scratchy and too loud when played by Thunderbird also sounded that way when played by VLC. Then I discovered several other .wav files on various Web sites that sounded distorted when played by the browser’s Windows Media Player plug-in (Gecko Media Player). So the problem clearly was not caused by Thunderbird itself. I began to wonder if PulseAudio was the cause. So I adjusted PulseAudio’s sampling frequency, number of fragments and fragment size, and all the clips that previously sounded distorted and too loud now play fine. Here is what I did to fix the problem…

1. Check what PulseAudio is configured to use:

$ pulseaudio --dump-conf
### Read from configuration file: /etc/pulse/daemon.conf ###
daemonize = no
fail = yes
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 5
allow-module-loading = yes
allow-exit = yes
use-pid-file = yes
system-instance = no
local-server-type = user
cpu-limit = no
enable-shm = yes
flat-volumes = no
lock-memory = no
exit-idle-time = 20
scache-idle-time = 20
dl-search-path = /usr/lib64/pulse-5.0/modules
default-script-file = /etc/pulse/default.pa
load-default-script-file = yes
log-target =
log-level = notice
resample-method = auto
enable-remixing = yes
enable-lfe-remixing = no
default-sample-format = s16le
default-sample-rate = 44100
alternate-sample-rate = 48000
default-sample-channels = 2
default-channel-map = front-left,front-right
default-fragments = 4
default-fragment-size-msec = 25
enable-deferred-volume = yes
deferred-volume-safety-margin-usec = 8000
deferred-volume-extra-delay-usec = 0
shm-size-bytes = 0
log-meta = no
log-time = no
log-backtrace = 0
rlimit-fsize = -1
rlimit-data = -1
rlimit-stack = -1
rlimit-core = -1
rlimit-rss = -1
rlimit-as = -1
rlimit-nproc = -1
rlimit-nofile = 256
rlimit-memlock = -1
rlimit-locks = -1
rlimit-sigpending = -1
rlimit-msgqueue = -1
rlimit-nice = 31
rlimit-rtprio = 9
rlimit-rttime = 1000000
$

2. Check PulseAudio’s output sample rate:

$ pacmd list-sinks | grep sample
sample spec: s16le 2ch 44100Hz
$

So the sample rate is 16 bits @ 44100 Hz and there are 2 output channels (Stereo). My laptop does indeed have two built-in stereo speakers. ‘s16le‘ means ‘signed 16-bit little-endian’.

3. Check PulseAudio’s input sample rate:

$ pacmd list-sources | grep sample
sample spec: s16le 2ch 44100Hz
sample spec: s16le 2ch 44100Hz
$

So the sample rate is 16 bits @ 44100 Hz and there are 2 input channels (Stereo). My laptop does indeed have two built-in microphones. ‘s16le‘ means ‘signed 16-bit little-endian’.

4. Find out the audio card’s maximum sample rate (Hz):

$ arecord -f dat -r 60000 -D hw:0,0 -d 5 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 60000 Hz, Stereo
Warning: rate is not accurate (requested = 60000Hz, got = 48000Hz)
please, try the plug plugin
$

5. As the console output shows that my audio card supports a sample rate of 48000 Hz, edit /etc/pulse/daemon.conf and change the sample rate accordingly:

$ su
Password:
# grep sample-rate /etc/pulse/daemon.conf
; default-sample-rate = 44100
; alternate-sample-rate = 48000
# nano /etc/pulse/daemon.conf
# grep sample-rate /etc/pulse/daemon.conf
default-sample-rate = 48000
; alternate-sample-rate = 48000
# exit
exit
$

6. Find the buffer size and fragment size for each sound card:

$ echo autospawn = no >> ~/.config/pulse/client.conf
$ pulseaudio --kill
$ LANG=C timeout --foreground -k 10 -s kill 10 pulseaudio -vvvv 2>&1 | grep device.buffering -B 10
I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] sink.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] sink.c: device.bus = "pci"
I: [pulseaudio] sink.c: device.vendor.id = "8086"
I: [pulseaudio] sink.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] sink.c: device.product.id = "3b56"
I: [pulseaudio] sink.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] sink.c: device.form_factor = "internal"
I: [pulseaudio] sink.c: device.string = "front:0"
I: [pulseaudio] sink.c: device.buffering.buffer_size = "19200"
I: [pulseaudio] sink.c: device.buffering.fragment_size = "4800"
--
I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] source.c: device.bus = "pci"
I: [pulseaudio] source.c: device.vendor.id = "8086"
I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] source.c: device.product.id = "3b56"
I: [pulseaudio] source.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] source.c: device.form_factor = "internal"
I: [pulseaudio] source.c: device.string = "front:0"
I: [pulseaudio] source.c: device.buffering.buffer_size = "19200"
I: [pulseaudio] source.c: device.buffering.fragment_size = "4800"
$ sed -i '$d' ~/.config/pulse/client.conf # You can just delete the file instead if it didn't exist in the first place.
$

N.B. Depending on your distribution, the PulseAudio file client.conf (if it exists) may be in a different sub-directory of the user’s home directory.

The console output shows that the buffer size was 19200 bits and the fragment size was 4800 bits.

7. Calculate the number of fragments and the fragment size (msec):

default-fragments = 19200 / 4800 = 4

default-fragments-size-msec

= device.buffering.fragment_size [bits] / (sample rate [Hz] x sample width [bits] x number of channels)

= 4800 / ( 48000 x 16 x 2 )

= 0.003125 seconds = 3.125 msec = 3 msec to the nearest integer

8. Edit /etc/pulse/daemon.conf to set these two parameters to the above-mentioned values:

# grep default-fragment /etc/pulse/daemon.conf
; default-fragments = 4
; default-fragment-size-msec = 25
# nano /etc/pulse/daemon.conf
# grep default-fragment /etc/pulse/daemon.conf
default-fragments = 4
default-fragment-size-msec = 3
# exit
exit
$

9. Restart PulseAudio:

$ pulseaudio --kill
$ pulseaudio --start # Only needed if you have 'autospawn = no' in ~/.config/pulse/client.conf
$

10. Check the PulseAudio configuration:

$ pulseaudio --dump-conf
### Read from configuration file: /etc/pulse/daemon.conf ###
daemonize = no
fail = yes
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 5
allow-module-loading = yes
allow-exit = yes
use-pid-file = yes
system-instance = no
local-server-type = user
cpu-limit = no
enable-shm = yes
flat-volumes = no
lock-memory = no
exit-idle-time = 20
scache-idle-time = 20
dl-search-path = /usr/lib64/pulse-5.0/modules
default-script-file = /etc/pulse/default.pa
load-default-script-file = yes
log-target =
log-level = notice
resample-method = auto
enable-remixing = yes
enable-lfe-remixing = no
default-sample-format = s16le
default-sample-rate = 48000
alternate-sample-rate = 48000
default-sample-channels = 2
default-channel-map = front-left,front-right
default-fragments = 4
default-fragment-size-msec = 3
enable-deferred-volume = yes
deferred-volume-safety-margin-usec = 8000
deferred-volume-extra-delay-usec = 0
shm-size-bytes = 0
log-meta = no
log-time = no
log-backtrace = 0
rlimit-fsize = -1
rlimit-data = -1
rlimit-stack = -1
rlimit-core = -1
rlimit-rss = -1
rlimit-as = -1
rlimit-nproc = -1
rlimit-nofile = 256
rlimit-memlock = -1
rlimit-locks = -1
rlimit-sigpending = -1
rlimit-msgqueue = -1
rlimit-nice = 31
rlimit-rtprio = 9
rlimit-rttime = 1000000
$

Notice that PulseAudio is now configured to use new values for default-sample-rate, default-fragments and default-fragment-size-msec.

$ echo autospawn = no >> ~/.config/pulse/client.conf
$ pulseaudio --kill
$ LANG=C timeout --foreground -k 10 -s kill 10 pulseaudio -vvvv 2>&1 | grep device.buffering -B 10
I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] sink.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] sink.c: device.bus = "pci"
I: [pulseaudio] sink.c: device.vendor.id = "8086"
I: [pulseaudio] sink.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] sink.c: device.product.id = "3b56"
I: [pulseaudio] sink.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] sink.c: device.form_factor = "internal"
I: [pulseaudio] sink.c: device.string = "front:0"
I: [pulseaudio] sink.c: device.buffering.buffer_size = "2304"
I: [pulseaudio] sink.c: device.buffering.fragment_size = "576"
--
I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] source.c: device.bus = "pci"
I: [pulseaudio] source.c: device.vendor.id = "8086"
I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] source.c: device.product.id = "3b56"
I: [pulseaudio] source.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] source.c: device.form_factor = "internal"
I: [pulseaudio] source.c: device.string = "front:0"
I: [pulseaudio] source.c: device.buffering.buffer_size = "2304"
I: [pulseaudio] source.c: device.buffering.fragment_size = "576"
$ rm ~/.config/pulse/client.conf # I didn't have a client.conf to begin with, so I just deleted it.
$ pulseaudio --start
$

Notice that PulseAudio is now using new values for device.buffering.buffer_size and device.buffering.fragment_size.

11. Check PulseAudio’s output sample rate:

$ pacmd list-sinks | grep sample
sample spec: s16le 2ch 48000Hz
$

Notice that PulseAudio is now using a new output sample rate.

12. Check PulseAudio’s input sample rate:

$ pacmd list-sources | grep sample
sample spec: s16le 2ch 48000Hz
sample spec: s16le 2ch 48000Hz
$

Notice that PulseAudio is now using a new input sample rate.

About Fitzcarraldo
A Linux user with an interest in all things technical.

3 Responses to Audio in Linux becomes annoying again (continued)

  1. svim says:

    Over the past few years every time I read some story about PulseAudio I’m glad I’m an old-school Linux distro user. Slackware by default doesn’t include PulseAudio and even on various desktops and laptops I’ve been happy with just alsa.

  2. Mat says:

    Thank you very much for this article, it helped me a lot !

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