Subject: JAPA bug ###################################################################### Date: Fri, 16 Aug 2024 04:56:47 +0000 From: Juan P C To: "fons@linuxaudio.org" http://kokkinizita.linuxaudio.org/linuxaudio/index.html -A -C pulse does not allow to select number of channels. Audacity works ok. ###################################################################### Date: Fri, 16 Aug 2024 09:11:24 +0200 From fons@linuxaudio.org Fri Aug 16 09:11:24 2024 From: Fons Adriaensen To: Juan P C On Fri, Aug 16, 2024 at 04:56:47AM +0000, Juan P C wrote: > -A -C pulse > does not allow to select number of channels. Nor does it make Coffee. Because it isn't supposed to do either. -C selects an ALSA input device. Try japa -h Ciao, -- FA ###################################################################### Date: Sat, 17 Aug 2024 02:08:13 +0000 From: Juan P C To: Fons Adriaensen Problem1: Display is limited to 20Khz, running at 96KHz Display should be 48KHz $ japa -A -d hw:2 -r 96000 -p 1024 -n 2 playback : not enabled capture : nchan : 14 fsamp : 96000 fsize : 1024 nfrag : 2 format : S32_LE Connected to ALSA with 14 inputs and 0 outputs Can't create ALSA thread with RT priority ---------------------------------- Problem2: to Analyze a Youtube video, or Odysee, or Rumble, or Bitchute, Vimeo, Dailymontion, etc... Pulse Audio has virtual cable/device to all audio outs, ALSA is HW: only, No SW:0 Audacity Records pulse ok. preferences: Host: ALSA Device: pulse Channels: 2 Japa has 0 channels s8 bit using device pulse but detects device "pulse" "Pulse" Not detected. $ japa -A -C pulse -r 96000 -p 1024 -n 2 playback : not enabled capture : nchan : 0 fsamp : 96000 fsize : 1024 nfrag : 2 format : S8 Connected to ALSA with 0 inputs and 0 outputs Can't create ALSA thread with RT priority -------------------------------------- $ japa -A -C pulse -P pulse -r 48000 -p 1024 -n 2 playback : nchan : 0 fsamp : 48000 fsize : 1024 nfrag : 2 format : S8 capture : nchan : 0 fsamp : 48000 fsize : 1024 nfrag : 2 format : S8 not synced Connected to ALSA with 0 inputs and 0 outputs Can't create ALSA thread with RT priority ---------------------------------- $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: CS4208 Analog [CS4208 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 1: CS4208 Digital [CS4208 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: USB [Scarlett 18i20 USB], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 ------------------------------- Problem3: i could Not subscribe to mail list... confirmation code gives error. -dev and the other mail list https://lists.linuxaudio.org/postorius/lists/ please open a Github or Gitlab. ###################################################################### Date: Sat, 17 Aug 2024 09:42:55 +0200 From fons@linuxaudio.org Sat Aug 17 09:42:55 2024 From: Fons Adriaensen To: Juan P C On Sat, Aug 17, 2024 at 02:08:13AM +0000, Juan P C wrote: > Problem1: Display is limited to 20Khz, > running at 96KHz Display should be 48KHz No. Japa is a 'perceptual' analyser, it makes no sense to go above 20 kHz. If you want to do 'technical' measurements, use jaaa, it will go up to half the sample rate. > Problem2: > preferences: Host: ALSA Device: pulse - Apparently this 'pulse' device doesn't provide any 16, 24, or 32 bit sample formats, only 8 bit signed (S8). That's pretty useless so it is not accepted. - Or the 'pulse' device is not correctly coded as an ALSA device. It wouldn't be the first one. If you want to do anything serious use Jack instead of pulseaudio, then you can connect anything to anything. You can even combine Jack and PA. > Problem3: i could Not subscribe to mail list... > confirmation code gives error. No idea why, but it could be this: when the confirmation code is longer than a single text line in your mail reader and you copy/paste it into a browser, it may include one or more newline characters. You'll need to remove those. > please open a Github or Gitlab. I do not support shared development of my code. If you report a real bug it will get my attention. Ciao, -- FA ###################################################################### Date: Sat, 17 Aug 2024 17:38:05 +0000 From: Juan P C To: Fons Adriaensen PulseAudio does Not run outside configuration parameters. default is s16le 44.1KHz needs to be manually configured for S24 & 96KHz $ cat /etc/pulse/daemon.conf has an optional resampler SRC when file/source is outside SR parameters. i have ffmpeg float set if file is 24-bit, and default 16 audio is truncated to 16. if file is 48KHz or 96khz, audio is resampled by SRC algorithm to configuration setting 1 or 2. if configuration 1 & 2 is 44.1KHz, output is 44.1 no matter what the input is. same bit depth, if set to s16le, is s16le. ###################################################################### Date: Sat, 17 Aug 2024 20:36:56 +0200 From fons@linuxaudio.org Sat Aug 17 20:36:56 2024 From: Fons Adriaensen To: Juan P C On Sat, Aug 17, 2024 at 05:38:05PM +0000, Juan P C wrote: > PulseAudio does Not run outside configuration parameters. > > default is s16le 44.1KHz > > needs to be manually configured for S24 & 96KHz > $ cat /etc/pulse/daemon.conf > > has an optional resampler SRC when file/source is outside SR parameters. > i have ffmpeg float set > > if file is 24-bit, and default 16 > audio is truncated to 16. > > if file is 48KHz or 96khz, audio is resampled by SRC algorithm to configuration setting 1 or 2. > if configuration 1 & 2 is 44.1KHz, output is 44.1 no matter what the input is. > same bit depth, if set to s16le, is s16le. What have PulseAudio and ffmpeg to do with japa ? Answer: nothing. -- FA ###################################################################### Date: Sat, 17 Aug 2024 19:35:27 +0000 From: Juan P C To: Fons Adriaensen im just trying to help you, improve your software. i assume you want to improve your software, & PulseAudio is the Default Audio System in All Linux Distros. a hub for audio, allows to route inputs & outputs. JAPA/JAAA requires inputs ALSA hw:0 limits JAPA/JAAA to Hardware only inputs. if someone want to analyze an audio file from VLC, or from youtube, without connecting real wires. PulseAudio is the best method. Qjackctl works amazing, but needs to open another software, close previous audio software "Firefox", open again, configure jack, run, connect virtual wires, save presets, if you have many tabs open, its time consuming.... interrupts the workflow. for complex routings: Qjackctl + Carla is prefered, for simple things: PulseAudio is prefered. there are softwares Only Jack, for example: $ jkmeter -rate 60 -chan 6 -C -V there is No other option, but i use less jkmeter, because its time consuming to run. the more latency you have, your brain gets disconnected from ideas. if the mouse & monitor had more latency, nobody will use computers. its simple logic. if this email took 6 months to arrive... this conversation would require 5 years. im just pointing out the obvious. i dont know why you do not want to improve your software, makes No sense to me. ###################################################################### Date: Sat, 17 Aug 2024 23:03:18 +0200 From fons@linuxaudio.org Sat Aug 17 23:03:18 2024 From: Fons Adriaensen To: Juan P C On Sat, Aug 17, 2024 at 07:35:27PM +0000, Juan P C wrote: > Qjackctl works amazing, but needs to open another software, > close previous audio software "Firefox", open again, > configure jack, run, connect virtual wires, save presets, > if you have many tabs open, its time consuming.... > interrupts the workflow. That's nonsense. PulseAudio is meant for desktop 'entertainment' audio. It is completely useless for anything else and not even installed on any of my systems. Here with a sinlge click I start Jack and some other apps (an audio file player and zita-mu1) when I login, and they remain running until I log out. I can easily do for example all the following at the same time: - Watch a YT video. - Play an audio file and visualise it with japa. - Record an external audio signal with Ardour. - Do a VOIP telephone call. - Send or receive audio from another computer on the local network. - etc. etc. All these things connect via Jack and can be stopped and started without interfering with each other. With a single click on the zita-mu1 input selection buttons I can switch listening between all these signals. And all this is of course just an example. There is never a need to stop, reconfigure and restart Jack. Just start the application you want to use and connect it to where you want. Or have it autoconnect if that makes sense. Japa, jaaa, jnoisemeter and some others are technical apps, meant to do accurate and reliable measurements. PA is by design absolutely useless for such work so I do not and never will support it. If you have a real bug report or suggestion to improve any of my applications, you are welcome. But don't try to teach me how to run a professional audio setup on Linux - I've more experience with that than you can dream of. -- FA ###################################################################### im Not teaching you, im showing my Point of View, your workflow does Not Fit everybody, & every scenario. i use Qjackctl for low latency audio... RT, with Liquorix Kernel, 256 Buffer x2 or 128 x4 / x5 my CPU cannot go lower without Xruns. but Liquorix kernel is Not stable with Firefox, has memory crash with 8GB machine, using lowlatency kernel + small Buffer has Xruns with FireFox + 20 Tabs open. generic kernel is useless for low latency audio. Firefox does Not have Jack output by default, needs to be compiled with options, or use a plugin to convert PulseAudio to Jack. same as VLC to Stop Qjackctl, change buffer size... reboot machine, grub different Kernel, close all 20 tabs, makes No sense. if you Run Qtjack at 1024 buffer, thats your scenario, but to me, 1024 makes No sense to me makes sense to run PulseAudio at 1024 buffer and Jack for Low Latency RT your software does Not require Low Latency Buffer Nor RT, instead: $ sudo xwax its absurd to run Xwax with 1024 Buffer or more. Carla + Virtual Synths the same, 1024 makes No sense. Large Buffer in CPU is for Mixdown, Not to Play RT. if you change too much settings in jack, sometimes needs reboot or logout. you are so close to make a perfect software, thats why i continue to write, jkmeter is so far its pointless to ask. P.D. The option to change Day & Night dark Skin mode would be awesome. like Voxengo plugins. ###################################################################### Date: Sun, 18 Aug 2024 10:41:14 +0200 From fons@linuxaudio.org Sun Aug 18 10:41:14 2024 From: Fons Adriaensen To: Juan P C On Sun, Aug 18, 2024 at 01:52:41AM +0000, Juan P C wrote: > im Not teaching you, im showing my Point of View, I didn't ask for that, and we are not on a public forum. Almost everything you write shows your complete ignorance about Linux Audio. And your inability to write a single correctly structured sentence. The impression you leave is that of a 14 year old with serious problems growing up. Anyway you are banned from now, I won't see any more of your messages. -- FA ######################################################################