The waterfall continues to work, but there is no sound

[Andy R3MAV]

Hello, everyone
I'm trying to install a Phantom with an RSP1a receiver and a miri driver.
At first everything is fine, but then errors start pouring into the terminal:

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Sample rate is set to 2500000 Hz.
Tuned to 145000000 Hz.
Tuner gain set to 4.500000 dB.
Reading samples in async mode...
106436484 samples lost, 16384, 000000fe:06581882
Waterfall is sent every 2 FFTs
[2025-04-05 10:32:15] [connect] WebSocket Connection [::1]:43450 v13 "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_15_7) AppleWebKit/605.1.15 (KHTML, like Gecko) Version/16.6.1 Safari/605.1.15" /waterfall 101
[2025-04-05 10:32:57] [error] handle_read_frame error: websocketpp.transport:7 (End of File)
[2025-04-05 10:32:57] [error] handle_read_frame error: websocketpp.transport:7 (End of File)
[2025-04-05 10:33:19] [error] handle_read_frame error: websocketpp.transport:7 (End of File)
[2025-04-05 10:33:19] [error] handle_read_frame error: websocketpp.transport:7 (End of File)
[2025-04-05 10:33:19] [info] asio async_write error: system:32 (Broken pipe)
[2025-04-05 10:33:19] [fatal] handle_write_frame error: websocketpp.transport:2 (Underlying Transport Error)


And on the client's side, the sound disappears. The waterfall continues to work, but there is no sound.

If you have the opportunity, look at the effect directly on my website:
http:websdr.srr-76.ru:8905

[Bas ON5HB]

I never got that to work, better use Soapy with the 2.13 API and rxtools as described somewhere in the forum.

[Andy R3MAV]

I installed the recommended programs, but the sound disappeared completely. And I can't adjust the RSP1A gain, I don't need an AGC, but without it, the waterfall is empty. But the most important thing is the lack of sound. I'm starting to get frustrated with this program.

That's how it is now. All default gain settings:

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rx_sdr -f 145000000  -s 2500000  -d driver=sdrplay  -t rfnotch_ctrl=true,dabnotch_ctrlb=true -F CS16 - | ./build/spectrumserver --config config-rsp1a.toml  &

By the way, it turned out that the loss of sound in the frontend does not depend on the device. I tried to run rtlsdr and RSP1a, the result is almost the same. Maybe the reason is that I don't use acceleration? Which web servers do you use, I have apache.

I don't even know what to think anymore...

[Bas ON5HB]

Normally the websever is build in the program.
Nobody uses apache.

Also, what version of the software do you use? Community or Original?

There are 3 versions....

We all use this version:

https://github.com/ny4qphil/PhantomSDR-Plus

As for the samplerate, I do not believe SDRplay supports 2.5MSPS.

Apache is not needed or supported by the software.

It runs directly on the port you gave in the TOML file.
Also posting the TOML file will help a lot.

[Andy R3MAV]

I'm using the community version. I didn't know about the embedded server at all, now I'm going to try, now. I'll report the results later.

[Andy R3MAV]

My config-rsp1a.toml

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[server]
port=8080# Server port
html_root="frontend/dist/" # HTML files to be hosted
otherusers=1 # Send where other users are listening, 0 to disable
threads=10

[websdr]
register_online=false # If the SDR should be registered on https://sdr-list.xyz then put it to true
name="Yaroslav VHF WebSDR (test)" # Name that is shown on https://sdr-list.xyz
antenna="Vertical" # Antenna that is shown on https://sdr-list.xyz
grid_locator=" LO07AQ" # 4 or 6 length Grid Locatlr shown on https://sdr-list.xyz and for the Distance of FT8 Signals
hostname="" # If you use ddns or something to host with a domain enter it here for https://sdr-list.xyz

[limits]
audio=100
waterfall=100
events=100

[input]
sps=3000000 # Input Sample Rate
fft_size=1048576
fft_threads=1
brightness_offset=1 # Waterfall brightness offset. Reduce to negative if you see black regions in the waterfall
frequency=145000000 # Baseband frequency
signal="iq" # real or iq
accelerator="none" # Accelerator: none, cuda, opencl
audio_sps=44100 # Audio Sample Rate
smeter_offset=-15
waterfall_size=2048
waterfall_compression="zstd"
audio_compression="flac" # flac or opus

[input.driver]
name="stdin" # Driver name
format="s16" # Sample format: u8, s8, u16, s16, u32, s32, f32, f64

[input.defaults]
frequency=145350000 # Default frequency to show user
modulation="FM" # Default modulation


You really don't need a web server. But the problem has not disappeared. It is only necessary to change the tuning frequency or modulation, and the sound disappears. The rest of the interface works fine, but there will be no sound until the page is reloaded in the browser.

After many experiments, it became clear that the problem was related to FM modulation. If you choose any other one, everything works stably. Immediately after switching to FM, the sound disappears.

There are no labels on the buttons Static IF Filter 6 kHz and 9kHz yet

[Bas ON5HB]

Threads for the server doesn't need that mucht, only 1 or 2 is enough.

As for audio_sps, it's recommended you use 12000 only, else e.g. FT-8 decoding won't work.

fft_treads you may want to use 2 or more as you don't use a GPU for acceleration.

And the samplerate sps isn't consistant with the rate you gave at rx_sdr, they must be the same.

RSP1A supports:

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Sample rates: 0.25, 0.5, 1, 2, 2.048, 6, 7, 8, 9, 10 MSps

RSP1A, you can not use any samplerate, but beware, the higher it's set the more CPU power you need.

You really want to use the Intel GPU to accelerate matters.

FM is far more CPU intensive than AM/SSB/CW.