Just be warned that speex-float-0 is the fastest and lowest quality from the speex-float-? resamplers. The default is speex-float-1 and I'd say that you could probably go a little higher with little cpu usage increase. Right now I'm using the default(1).
My problem with skype on the other thread turned out to be related to something else than the resampler but that was the workaround I found at the time. The problem was that PULSE_LATENCY_MSEC had to be set, that seems to be a known bug upstream, however it might also apply to other apps that output to pulse directly.
(1) Now I can't detect any artifacts when pulse needs to resample audio (maybe also because pulse can now switch to an alternate sampling rate) but a (long) while ago the default resampler (some speex-float) introduced quite noticeable artifacts, going to a speex-float resampler that was ok required too much cpu time, so I fiddled with the resampler and found that ffmpeg was a nice compromise. Since you had src-sinc-best-quality as the resampler you probably care about quality so do try other resamplers and see what works best for you.
You are right. Since I've got decent speakers (and prefer using FLAC over mp3), I try to achieve good quality. I think, I'll stick to speex-float-7 for now.
Thank you
My problem with skype on the other thread turned out to be related to something else than the resampler but that was the workaround I found at the time. The problem was that PULSE_LATENCY_MSEC had to be set, that seems to be a known bug upstream, however it might also apply to other apps that output to pulse directly.
(1) Now I can't detect any artifacts when pulse needs to resample audio (maybe also because pulse can now switch to an alternate sampling rate) but a (long) while ago the default resampler (some speex-float) introduced quite noticeable artifacts, going to a speex-float resampler that was ok required too much cpu time, so I fiddled with the resampler and found that ffmpeg was a nice compromise. Since you had src-sinc-best-quality as the resampler you probably care about quality so do try other resamplers and see what works best for you.
]]>Try changing the resample method, you have changed it from the default so try with the default first.
Thanks to all of you.
This one fixed it.. Changing resample-method back to
resample-method=speex-float-0
did the trick!
]]>can you try pulseaudio's simultaneous output.
pacman -S paprefs
open paprefs, go to Simultanous Output tab and enable it
then go to your volume control and change the output to Simultaneous and check if the sounds still cracklingalso would be nice if you post your /etc/pulse/daemon.conf
I tried paprefs with the simultaneous output enabled and it was still doing that.
These are the contents of my daemon.conf:
; daemonize = no
; fail = yes
; allow-module-loading = yes
; allow-exit = yes
; use-pid-file = yes
; system-instance = no
; local-server-type = user
; enable-shm = yes
; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
; lock-memory = no
; cpu-limit = no
; high-priority = yes
; nice-level = -11
; realtime-scheduling = yes
; realtime-priority = 5
exit-idle-time=0
; exit-idle-time = 20
; scache-idle-time = 20
; dl-search-path = (depends on architecture)
; load-default-script-file = yes
; default-script-file = /etc/pulse/default.pa
; log-target = auto
; log-level = notice
; log-meta = no
; log-time = no
; log-backtrace = 0
resample-method=src-sinc-best-quality
; resample-method=speex-float-0
; resample-method = speex-float-3
; enable-remixing = yes
; enable-lfe-remixing = no
; flat-volumes = yes
; rlimit-fsize = -1
; rlimit-data = -1
; rlimit-stack = -1
; rlimit-core = -1
; rlimit-as = -1
; rlimit-rss = -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
; 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
That all looks rational. Next, I would fire up pavucontrol and ensure that all the input sources are muted (like microphones). Flip over to the configuration tab and try some different configurations.
You might also post the output of pulseaudio --dump-conf and I'll compare it to mine.
Again, I've no solid ideas, I am still just probing.
I did not have pavucontrol installed.. I did so, muted the mic and tried some different settings.. but I still got no success.
As far as I saw, that output is the content of my daemon.conf which I postet above in this post.
No problem.. I'm glad you're trying to help.
]]>You might also post the output of pulseaudio --dump-conf and I'll compare it to mine.
Again, I've no solid ideas, I am still just probing.
]]>pacman -S paprefs
open paprefs, go to Simultanous Output tab and enable it
then go to your volume control and change the output to Simultaneous and check if the sounds still crackling
also would be nice if you post your /etc/pulse/daemon.conf
]]>null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default
Default ALSA Output (currently PulseAudio Sound Server)
sysdefault:CARD=PCH
HDA Intel PCH, ALC892 Analog
Default Audio Device
front:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
Front speakers
surround40:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Digital
IEC958 (S/PDIF) Digital Audio Output
hdmi:CARD=NVidia,DEV=0
HDA NVidia, HDMI 0
HDMI Audio Output
hdmi:CARD=NVidia,DEV=1
HDA NVidia, HDMI 0
HDMI Audio Output
hdmi:CARD=NVidia,DEV=2
HDA NVidia, HDMI 0
HDMI Audio Output
hdmi:CARD=NVidia,DEV=3
HDA NVidia, HDMI 0
HDMI Audio Output
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 7: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 8: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 9: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
Codecs.. I got most of the common ones installed, if that is what you want to know.
Not quite. I meant hardware codecs.
What is the output of aplay -lL (that is a lower case L followed by an upper case L)
I've noticed this, when I was playing some Video on youtube and a gnome notification appeared.. instead of the usual tone, there was this ugly crackling. This also happens when I'm playing an audio file on some player and the gnome notification appears.. If I'm trying the gnome speaker testing thing while playing something, it also results in a crackling noise.
I just noticed that it does NOT happen, when I play something on youtube and an audio file at the same time.
load-module module-udev-detect tsched=0
in my /etc/pulse/default.pa but this does not seem to have any effect.
Hope you can help me..
]]>