Hello asahi linux team, I use asahi regularly for audio production on my macbook air m2 (j413 speakers) and don't bypass any of your processing filters to take advantage of all of the work you have done to make the audio sound good. Recently I have tried to use the roland sc55 emulator, specifically this fork: https://github.com/jcmoyer/Nuked-SC55 together with my DAW Radium. By default it uses strange sample rates to match the hardware and outputs 66207hz, which pipewire correctly provides, this fork specifically also has a flag to disable oversampling --disable-oversampling, halving the frequency to 33103hz. Running pw-top, pipewire reports 44100hz while the biggest common sample rate is 48000hz, and the application insists it got 33103hz:
Base path is: /home/leo/Coding/Nuked-SC55/build
ROM directory is: /home/leo/Coding/Nuked-SC55/build
Using SC-55mk2 romset:
* ROM1 /home/leo/Coding/Nuked-SC55/build/r15199858_main_mcu.bin
* ROM2 /home/leo/Coding/Nuked-SC55/build/r00233567_control.bin
* SMROM /home/leo/Coding/Nuked-SC55/build/r15199880_secondary_mcu.bin
* WAVEROM1 /home/leo/Coding/Nuked-SC55/build/r15209359_pcm_1.bin
* WAVEROM2 /home/leo/Coding/Nuked-SC55/build/r15279813_pcm_2.bin
WARNING: No reset specified with mk2 romset; using gs
Gain set to 0.00db
Audio device: Default device (SDL)
Audio requested: format=AUDIO_S16LSB, channels=2, frequency=33103, frames=512
Audio actual: format=AUDIO_S16LSB, channels=2, frequency=33103, frames=512
#00: allocated 65536 bytes for audio
Opened midi port: Midi Through:Midi Through Port-0 14:0
Unknown write e400 4
Unknown write e403 1
Unknown write e406 0
Unknown write e407 0
Unknown write e406 40
Unknown write e403 1
pw-top output before routing to radium:
S ID QUANT RATE WAIT BUSY W/Q B/Q ERR FORMAT NAME
I 30 0 0 0.0us 0.0us ??? ??? 0 Dummy-Driver
S 31 0 0 --- --- --- --- 0 Freewheel-Driver
S 45 0 0 --- --- --- --- 0 Midi-Bridge
S 48 0 0 --- --- --- --- 0 bluez_midi.server
R 55 1024 48000 1.2ms 0.3us 0.06 0.00 2 S32LE 1 48000 alsa_input.platform-sound.HiFi__Headset__source
R 54 0 0 15.9us 62.2us 0.00 0.00 2 S32LE 2 48000 + alsa_output.platform-sound.HiFi__Headphones__sin
R 113 0 0 36.9us 1.1ms 0.00 0.05 0 + radium_audio
I 60 0 0 0.0us 0.0us ??? ??? 0 F32P 3 0 audio_effect.j413-mic
S 61 0 0 --- --- --- --- 0 effect_output.j413-mic
R 71 0 0 0.0us 0.0us ??? ??? 0 F32P 2 0 audio_effect.j413-convolver
R 72 0 0 0.0us 0.0us ??? ??? 0 F32P 4 0 effect_output.j413-convolver
S 105 0 0 --- --- --- --- 0 v4l2_input.platform-22a000000.isp
R 121 0 0 0.0us 0.0us ??? ??? 0 S16LE 2 44100 SDL Application
I am routing the output of nuked manually through my daw using helvum, which correctly outputs to the speaker using pipewire jack audio connection kit letting it perform all the dsp. Therefore is there any intermediate processing that upsamples the audio. The convolver maybe? Assuming there is, is there a way for me to bypass that processing for this specific application only when launching it, so that I can route it trough my DAW at the correct sample rate where it is correctly processed? If there is no other intermediate processing then I'd greatly appreciate if you can tell me or if we can figure out what the problem is.
Hello asahi linux team, I use asahi regularly for audio production on my macbook air m2 (j413 speakers) and don't bypass any of your processing filters to take advantage of all of the work you have done to make the audio sound good. Recently I have tried to use the roland sc55 emulator, specifically this fork: https://github.com/jcmoyer/Nuked-SC55 together with my DAW Radium. By default it uses strange sample rates to match the hardware and outputs 66207hz, which pipewire correctly provides, this fork specifically also has a flag to disable oversampling
--disable-oversampling, halving the frequency to 33103hz. Runningpw-top, pipewire reports 44100hz while the biggest common sample rate is 48000hz, and the application insists it got 33103hz:pw-top output before routing to radium:
I am routing the output of nuked manually through my daw using helvum, which correctly outputs to the speaker using pipewire jack audio connection kit letting it perform all the dsp.
Therefore is there any intermediate processing that upsamples the audio. The convolver maybe? Assuming there is, is there a way for me to bypass that processing for this specific application only when launching it, so that I can route it trough my DAW at the correct sample rate where it is correctly processed? Ifthere is no other intermediate processing then I'd greatly appreciate if you can tell me or if we can figure out what the problem is.