The newer implementation of the PulseAudio sound server uses timer-based audio scheduling instead of the traditional, interrupt-driven approach. if you tested on Steam before modifying the file), in which case you should exit the application and manually start Pulseaudio Note: Pulseaudio may fail to start if you do not exit a program that was using the microphone (e.g. Load-module module-remap-source master=alsa_input.pci-0000_00_14.2.analog-stereo master_channel_map=front-left,front-right channels=2 channel_map=mono,mono Add a remap rule to /etc/pulse/default.pa, use the name you found with the previous command, here we will use alsa_input.pci-0000_00_14.2.analog-stereo as an example: To fix this you need to tell Pulseaudio to do this for you:Įxample output edited for brevity, the name you need is in bold:įlags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCYĢ. This happens because these applications capture the microphone as mono only and because remixing is disabled, Pulseaudio will no longer remix your stereo microphone to mono. When you set enable-remixing = no on /etc/pulse/nf you may find that your microphone has stopped working on certain applications like Skype or Steam. No microphone on Steam or Skype with enable-remixing = no You may still need to change the RecordStream from setting to Remapped Built-in Audio Analog Stereo of a specific application in the Recording tab of pavucontrol. Load-module module-remap-source source_name=record_mono master=INPUT_NAME master_channel_map=front-left channel_map=mono Edit /etc/pulse/default.pa and add the following lines, where INPUT_NAME is name of the input source from above step: Find your source name from the following command mine is alsa_input.pci-0000_00_1f.3.analog-stereoĢ. The solution is to remap the stereo input to a mono input:ġ. Now hopefully, there is no static noise in microphone recording anymore.Īnother possible cause is that your mic has two channels but only one channel can provide a valid sound signal. Make sure the microphone is not muted and allĪfter 10 seconds, let us play the recording. Let us record some voice using a microphone for, say, 10 seconds. Restart PulseAudio to apply the new settings (4/5)įinally check by recording and playing it back (5/5) # sed 's/ default-sample-rate = 48000/default-sample-rate = 44100/g' -i /etc/pulse/nf $ grep "default-sample-rate" /etc/pulse/nfĤ8000 is disabled and needs to be changed to 44100: Setting the sound card's sampling rate into PulseAudio configuration (3/5) This is the maximum sampling rate of our card. Warning: rate is not accurate (requested = 60000Hz, got = 44100Hz ) "Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 60000 Hz, Stereo When the top value is reached, we got a warning message:Īrecord -f dat -r 60000 -D hw:0,0 -d 5 test.wav We aim to find the highest sample rate supported by the hw:0,0 sound card using a trial-and-error procedure starting from a low value. In the above example, it is hw:0,0.ĭetermine sampling rate of the sound card (2/5) The sound card is hw: x, y where x is the card number and y is the device number. **** List of CAPTURE Hardware Devices ****Ĭard 0: Intel, device 0: ALC888 Analog Ĭard 0: Intel, device 2: ALC888 Analog This requires alsa-utils and related packages to be installed: ĭetermine sound cards in the system (1/5) In addition to the guide below, since PulseAudio 11 it is possible to set avoid-resampling = yes in nf. To fix this, we need to set the sampling rate in /etc/pulse/nf for the sound hardware. That is why there is static noise in Linux microphone recordings. If we are getting static noise in Skype, gnome-sound-recorder, arecord, etc.'s recordings, then the sound card sample rate is incorrect.
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