I want to use piper for text to voice. Am using Ubuntu 22.04 LTS and have installed it using the command sudo apt install piper Now running echo 'Hello world' | piper --model en_US-lessac-medium gives me Unknown option --model dpkg -l piper gave a short description of the package installed, "GTK application to configure gaming devices." I then installed piper-tts with pip, and restarted the machine. pip3 install piper-tts Running the command again echo 'Hello world' | piper Traceback (most recent call last): File "/home/hagbard/.local/lib/python3.11/site-packages/numpy/_core/__init__.py", line 22, in from . import multiarray File "/home/hagbard/.local/lib/python3.11/site-packages/numpy/_core/multiarray.py", line 11, in from . import _multiarray_umath, overrides ImportError: libz.so.1: cannot open shared object file: No such file or directory The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/hagbard/.local/lib/python3.11/site-packages/numpy/__init__.py", line 112, in from numpy.__config__ import show_config File "/home/hagbard/.local/lib/python3.11/site-packages/numpy/__config__.py", line 4, in from numpy._core._multiarray_umath import ( File "/home/hagbard/.local/lib/python3.11/site-packages/numpy/_core/__init__.py", line 48, in raise ImportError(msg) from exc ImportError: IMPORTANT: PLEASE READ THIS FOR ADVICE ON HOW TO SOLVE THIS ISSUE! Importing the numpy C-extensions failed. This error can happen for many reasons, often due to issues with your setup or how NumPy was installed. We have compiled some common reasons and troubleshooting tips at: https://numpy.org/devdocs/user/troubleshooting-importerror.html Please note and check the following: * The Python version is: Python3.11 from "/gnu/store/n8fxwzkq2bkg55p5s0x1largmmd1fyw4-python-3.11.11/bin/python3.11" * The NumPy version is: "2.3.2" and make sure that they are the versions you expect. Please carefully study the documentation linked above for further help. Original error was: libz.so.1: cannot open shared object file: No such file or directory The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/hagbard/.local/bin/piper", line 5, in from piper.__main__ import main File "/home/hagbard/.local/lib/python3.11/site-packages/piper/__init__.py", line 4, in from .voice import AudioChunk, PiperVoice File "/home/hagbard/.local/lib/python3.11/site-packages/piper/voice.py", line 13, in import numpy as np File "/home/hagbard/.local/lib/python3.11/site-packages/numpy/__init__.py", line 117, in raise ImportError(msg) from e ImportError: Error importing numpy: you should not try to import numpy from its source directory; please exit the numpy source tree, and relaunch your python interpreter from there.

I spend a lot of time taking pictures with a usb camera attached to a microscope. Thing is... I need to manipulate the plaque with my two hands, so moving one hand from the microscope to the computer to take a screenshot (space bar) is really difficult. Question is... Is there an app that hears my voice and takes pictures? I was thinking ... maybe a personal voice assistant is the way to go... but maybe an application (like gnome-cheese or something) already does the job.... My question is a bit like this one (https://unix.stackexchange.com/questions/616291/detect-simple-voice-commands) . I have to say 'cheese' or 'photo' or 'whatever' and that's gonna be the input for whatever program can take screenshots... I've been searching around and found this https://voice2json.org/ . Somewhere it says that > It can be used to: > > Add voice commands to existing applications or Unix-style workflows I do not know if this is overkill or very simple. Ps. I am using Fedora 35 Thanks

It seems easy to change voice in Windows just like this tutorial https://www.youtube.com/watch?v=pmlYiK6fgX0 Unfortunately, Voicemod is not available on Linux yet. I've been searching for Linux Voice Changer and found this link 7 Best Real Time Voice Changer for Discord Apps 2020 Out of 7 Best Real Time Voice Changer provided, only 2 supported by Linux which is Mumble and Team Speak I've installed both apps, unfortunately, I still can't figure out how to change a voice. Should there is any good tutorial on it, please let me know.

I've got a Huawei E173 USB dongle connected to a Raspberry PI. I'm trying to use the device to make and receive calls. For my purpose, it is needed (i) to save audio of the call to file and (ii) to send audio to the telephony call from an input file. As far as I understood (I can't find any official guide), the E173 dongle sets up three new devices (please correct me if I'm wrong): * /dev/ttyUSB0 for commands * /dev/ttyUSB1 for voice data * /dev/ttyUSB2 for notifications In order to enable phone calls, I run the following commands: AT+CLIP=1, AT+CRC=1, AT+CNMI=1,2 on /dev/ttyUSB0. Also, if I run AT^CVOICE? I get ^CVOICE:0,8000,16,20. Finally, when I make/receive calls, I run AT^DDSETEX=2 on /dev/ttyUSB0 to enable audio forward to the /dev/ttyUSB1 port. At this point, I'm stuck. I know there are other similar posts trying to redirect microphone input and speakers output to the dongle, but my aim is to save the conversation to file (during a call, I tried to run cat /dev/ttyUSB1 > filename.raw, but the file can't be opened with standard software, hence, how can I convert it?) and, simultaneously, to send voice from file (which format? I tried to send back the filename.raw to /dev/ttyUSB1 with cat filename.raw > /dev/ttyUSB1, but the entire system gets freezed). How can I save audio to file and send back audio from file? I found some resources on the Internet trying to explain, for similar devices, that audio has to be sent in mono, in digital frequency of 8000 Hz, and digitalized to 16 bit. By looking at the result of the AT^CVOICE? command, such parameters should be applied also in my case. Also, the same resource tells that "audio data should be fed to the modem audio port in batches of 320 bytes every 0.02 seconds". I've tried to apply even such approach, with no luck.

In the presence of current wave of COVID-19, we all are forced to work-from-home and attend online group meetings. In this context, my organization is currently using MS Teams for such meetings. However, we are trying to move away to some other similar product (as we are not comfortable to give control and data to MS). So, in this effort, first we tried Zoom, but it has a limitation of 40 minutes. Then we tried Jitsi, which is good but in Jitsi, anyone of the participants can kick-out any other (which is a serious problem). Therefore, we are still looking for an Open Source alternative to MS Teams at least following functionalities: > 1. Confrence Call > 2. Screen Sharing > 3. Only Admin can kick out any member If you happen to know any software with these features, kindly let me know. Thanks.

I have a Huawei E160E and a Huawei K3765 "GSM"-modem. The question is: How to send and receive audio data placing or receiving phone/ voice calls with those modems? Both seem to support voice calls, according to specifications (E160E , K3765 ) and according to AT-commands: Issuing AT^CVOICE? results in ^CVOICE:0,8000,16,20 OK (both modems the same). Both modems provide 3 USB ports, I will call them /dev/ttyUSB0, /dev/ttyUSB1, /dev/ttyUSB2 (assuming there is no other USB serial device present in the system). The K3765 additionally registers an cdc_etherethernet device. I am using Linux. On both /dev/ttyUSB0 and /dev/ttyUSB2 I can place AT-commands and receive their answers; additionally, /dev/ttyUSB2 reports status messages (like ^RSSI:23, ^BOOT:38021325,0,0,0,77, ^SIMST:1 or ^SRVST:2). The K3765 also reports by default when there is someone calling (RING), the E160E does not. On both modems, I can place calls with ATD; and I can pick up calls with ATA. /dev/ttyUSB1 seems to do nothing; simply writing to this file will get no data into it, reading from it gets no data out. How is it possible to get audio data in and out of the modem(s)? I already found this stackexchange question and the answer by Sai Chaitanya as well as the comment to use ffmpeg/ffplay. ffplay just stalls for me, and with the code snipped I can not make too much to try it out (not a programmer, and expecially not in C, though; and don't know if this applies to my modem(s) at all). In the internet I do not find information on how to actually get audio in and out of those devices, other than using big software packages (asterisk, for example) -- which I do not want, the question is on how to get it to work the basic way. So, how does audio communication work with (those) Huawei GSM modems? Modem information: Output of ATI-command: E160E: Manufacturer: huawei Model: E160X Revision: 11.609.10.02.432 IMEI: 354765038021325 +GCAP: +CGSM,+DS,+ES OK K3765: Manufacturer: huawei Model: K3765 Revision: 11.126.03.06.00 IMEI: 353142047899771 +GCAP: +CGSM,+DS,+ES OK

Only I want to install a Spanish Latin voice for Festival, then I was looking for them. I'm using CentOS, but I can't find the TTS packages... I'm not Expert in this matter. I was looking for using google, but the information I found is Ubuntu and .deb extension files. Other based on Asterisk and FestBox... RPM My question are: - Is FestVox needed? - Is Asterisk needed? - Where I can get Latin Spanish voices, please links... - Have you step by step installation of this voices? Is it possible to use another TTS (Loquendo, aCapela, etc) On Festival, are they interchangeable? Thank you!

I wish to create a simple two-way voice communication system between two Linux computers running Debian-based distros (probably two Raspberry Pi Zeroes). Each computer will have a microphone and speaker attached as well as a physical button. Both computers will be headless after setup and always-on. The computers would both be on the local network. Essentially, microphone input would need to be sent in real-time over the network by each computer and played by the other computer. The system would function as follows: When the button on either computer is pressed (let's say it is called Computer 1 in this case) Computer 1 will 'call' the other computer (Computer 2). Computer 2 will ring. When the button is pressed on Computer 2, the 'call' will be answered and voice communication will begin, with a person on each computer able to hear and speak to the other via the speakers and microphone. When either user presses the button on their computer, the 'call' will disconnect and voice communication will end, with the two computers going to their starting state, ready to accept a 'call' or make one when the button is pressed. How can I achieve this behaviour? Any solutions, including pointers towards relevant software are appreciated.

Audacity, for example, can be switched to JACK if needed. What about software that can't but can be switched to alsa?

I'm running a HelpDesk for internal call center employees. Right now, we can chat with employees over our web portal and remote-control their Linux Mint workstations via VNC over a secure SSH tunnel. All employees have headsets in conjunction with their jobs, and I'd like to find a way to initiate voice-chat with them over SSH in order to better aid in remote support. After some googling, I found two commands: arecord -f cd -t raw | oggenc - -r | ssh admin@REMOTEUSER mplayer - and dd if=/dev/dsp | ssh -c arcfour -C admin@REMOTEUSER dd of=/dev/dsp Neither of these commands (with proper credentials) seemed to work in my testing, however. Does anyone know how I can establish a live voice connection over SSH to an employee's machine? I have root access to the remote machines, so that won't be a problem. **UPDATE:** I tried hosting a Murmur server on my local network. Opening Mumble over SSH does work, but I can't seem to get Mumble to launch minimized (or without the GUI). Are there any other voice-chat solutions which I could host over LAN and hide the employee's chat GUI?

I have a set of audio files containing voice recordings, I need to alter the voice signal so the speaker cannot be recognised in an automated way...
*Do you know some application?*