Nixos no sound

[alkeryn]

So i finally succesfuly built it and have it work with my limesdr, it dosen't see my audio output devices tho so i don't have any sound, any idea why that is ? maybe some compilation flags or libs ?

There is no audio devices shown except the default one and when i click ok the software just crash and output this :

[f4exb]

That -1 Hz sample rate is strange. It should come from the config file normally and that may have been accidentally corrupted. You can try to erase ~/.config/f4exb/SDRangel.conf so it will rebuild it from scratch.

[alkeryn]

I tried removing it, changed nothing ^ Also i rebuild the whole thing with libav because it was missing, didn't fixed it ^ it just looks like that :

[f4exb]

What does the command pacmd list-sinks return?

[alkeryn]

❯ pacmd list-sinks
1 sink(s) available.
  * index: 0
    name: <alsa_output.pci-0000_00_1f.3.analog-stereo>
    driver: <module-alsa-card.c>
    flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
    state: RUNNING
    suspend cause: (none)
    priority: 9039
    volume: front-left: 18866 /  29% / -32,45 dB,   front-right: 18866 /  29% / -32,45 dB
            balance 0,00
    base volume: 65536 / 100% / 0,00 dB
    volume steps: 65537
    muted: no
    current latency: 178,66 ms
    max request: 36 KiB
    max rewind: 36 KiB
    monitor source: 2
    sample spec: s16le 2ch 44100Hz
    channel map: front-left,front-right
                 Stéréo
    used by: 1
    linked by: 1
    configured latency: 210,00 ms; range is 0,50 .. 371,52 ms
    card: 3 <alsa_card.pci-0000_00_1f.3>
    module: 9
    properties:
        alsa.resolution_bits = "16"
        device.api = "alsa"
        device.class = "sound"
        alsa.class = "generic"
        alsa.subclass = "generic-mix"
        alsa.name = "ALC1220 Analog"
        alsa.id = "ALC1220 Analog"
        alsa.subdevice = "0"
        alsa.subdevice_name = "subdevice #0"
        alsa.device = "0"
        alsa.card = "0"
        alsa.card_name = "HDA Intel PCH"
        alsa.long_card_name = "HDA Intel PCH at 0xde140000 irq 178"
        alsa.driver_name = "snd_hda_intel"
        device.bus_path = "pci-0000:00:1f.3"
        sysfs.path = "/devices/pci0000:00/0000:00:1f.3/sound/card0"
        device.bus = "pci"
        device.vendor.id = "8086"
        device.vendor.name = "Intel Corporation"
        device.product.id = "a2f0"
        device.product.name = "200 Series PCH HD Audio"
        device.form_factor = "internal"
        device.string = "front:0"
        device.buffering.buffer_size = "65536"
        device.buffering.fragment_size = "32768"
        device.access_mode = "mmap+timer"
        device.profile.name = "analog-stereo"
        device.profile.description = "Stéréo analogique"
        device.description = "Audio interne Stéréo analogique"
        alsa.mixer_name = "Realtek ALC1220"
        alsa.components = "HDA:10ec1220,18491221,00100003 HDA:8086280b,80860101,00100000"
        module-udev-detect.discovered = "1"
        device.icon_name = "audio-card-pci"
    ports:
        analog-output-lineout: Sortie ligne (priority 9900, latency offset 0 usec, available: yes)
            properties:

        analog-output-headphones: Casque audio (priority 9000, latency offset 0 usec, available: no)
            properties:
                device.icon_name = "audio-headphones"
    active port: <analog-output-lineout>

[f4exb]

So it appears that you do have pulseaudio and it should work but for some reason Qt does not recognize it. In the log of SDRangel near the start you should see a line like (you may need to turn on Debug level for the log):

...
2019-08-22 16:30:17.207 (D) AudioDeviceManager::AudioDeviceManager: input device #18: alsa_output.pci-0000_00_1f.3.analog-stereo

The number here is arbitrary the point is to see that pulseaudio device. I guess this is not the case since it should appear in the list in the GUI as well and as the system device.

[alkeryn]

the only line there is is this one ^ :

2019-08-22 15:40:32.105 (I) SDRangel 4.11.6 Qt 5.12.0 64b x86_64 NixOS 19.09pre186857.239fffc90d7 (Loris) DSP Rx:24b Tx:16b PID 24748

[f4exb]

Turn Debug on!

[alkeryn]

I did (if you mean logging debug in the ui) or is there a command line flag for debug ? or do you want me to compile with debug flag ?

I think it just don't see my audio devices

[alkeryn]

Oh that one is new :

2019-08-22 17:15:13.827 (W) AudioDeviceManager::setOutputDeviceInfo: index: -1 device: System default device no FIFO registered yet hence no audio output has been allocated yet

[f4exb]

The issue with debug is that Qt turns it off if you don't specify -DDEBUG_OUTPUT=ON

In the main CMakeLists.txt default is OFF:

option(DEBUG_OUTPUT "Print debug messages" OFF) 
...
if(NOT DEBUG_OUTPUT)
  add_definitions("-DQT_NO_DEBUG_OUTPUT")
endif()

So my guess is that you need that option on your cmake command line.

[alkeryn]

Oh ok, recompilling it ^

[alkeryn]

i don't think there is anything useful but here /shrug

[f4exb]

Strangely enough when debug is working you should see some messages between the program start at ~17:40 and the main window close event at ~17:41. It seems the debug flag was not taken into account before you closed the main window.

[alkeryn]

yeah, idk what i should do ^ it is weird, the only software that does that, i mean audacity or all other softwares have no such issue so maybe i'm forgetting to had a lib when building idk but dosen't look like it ^

[f4exb]

In my opinion if Qt has trouble finding the audio device then it has something to do with some Qt dependencies.

[alkeryn]

Hey i just fixed it ! it was indeed a qt issue and me being new to the nix programming language ^

[alkeryn]

Well i do still have some weird audio stutering and can't hear much but at least it detects my devices now ^

[alkeryn]

Oh it looks like it is just overloading like if i reduce S/S to 5M it works tho on gqrx i can go up to 25M/s and it don't stutter any recomendation for that ?

[f4exb]

Is this LimeSDR? I can go up to 10 MS/s without packet drops. It still goes beyond that point without guarantee of signal integrity.

Edit: if you mean audio stuttering you have to consider that audio runs at 48 kS/s so the system has to make the decimation from 25 MS/s to 48 kS/s. To make it work properly you have to share the load between the decimation for the device and for the channel with more decimation on the device. I would not exceed a few 100 kHz on the channel and get the rest from device decimation. For example: 25 MS/s decimating by at least 32 yields 781.25 kS/s which starts to be OK (you could use 64 as well). However you will only see 781.25 kHz of baseband. There is a choice to be made between narrowband work (except WFM and BFM all audio type plugins are narrowband) and wideband work.

[alkeryn]

Huh that's weird on gqrx i can go up to 30M/s Also weirdly enough disabling the top graph allow me to go up to 20M instead of 8 so it looks like it is cpu bound somehow :thinking: Yeah it is a limesdr