I have phone call recordings which are dual channel with each channel supposed to carry only the voice of one speaker. However, they have some echo of the other channel. Any ways to remove this, in ffmpeg or sox or otherwise.
I am working on a Ubuntu 16.04 environment and using mplayer to play back the audio. A link to a 10s clip of the audio may be found here: https://drive.google.com/file/d/14xrchHvcluhDNGutYfCPpQi3cas_4Ogi/view?usp=sharing
I also looked at (almost) the same question: Silence out quiet periods in audio file with ffmpeg
Not very sure I could follow the answer/comment though.
Thanks!
Related
I have a pipe module with the command pactl load-module module-pipe-source source_name=VirtualMic file=/tmp/virtualmic format=wav rate=44100 channels=2
I want to use SoX to play a sound file into it. I am doing this with sox "example.wav" -t wav - > /tmp/virtualmic
I have tried piping the audio using ffmpeg, to the same result, and to confirm that it is not my computer speakers or the file, playing the file in audio programs such as vlc does not include a popping sound.
The number of channels and the sample rate are both identical, and other then the pop the audio plays normally
My machine is running Ubuntu 20 LTS. I want to manipulate the input live audio in real-time. I have achieved pitch shifting using sox. The command being -
sox -t pulseaudio default -t pulseaudio null pitch +1000
and then routing the audio from "Monitor of Nullsink" .
What I actually want to do is, silence randomized parts of the input audio, with a range. What I mean is, randomly mute 1-2s of the input audio.
The final goal of this project will be to write a script that manipulates my voice and makes it seems like my network is bad.
There is no restriction in method of achieving. That is we may use any language, make an extension, directly manipulate the input audio with sox, ffmpeg etc. Anything goes.
Found the solution by using trim in sox. The project can be found in
https://github.com/TathagataRoy1278/Bad_Internet_Audio_Modulator
I'm using OpenMusic for the first time on Linux Mint 19.2; and I'm having issues getting any sound out of it from MIDI.
I have it set to Pulse Audio for output, and the PA Volume Control is open and above-other-windows, so I can see whether there's an output to it even if it's silenced. It's not muted, it's just dead silent. When I press the Play button on a note or measure, I still get nothing.
I've set MIDI out to timidity, and verified that it's set to use Pulse Audio. Am I missing something here, like a library or a sound font? How do I change that?
To get the obvious out of the way, my speakers are plugged in and on with the sound up; and other software has no issues with playback. I can play audio if it's from a file like an Ogg Vorbis, so this is likely MIDI related.
I have an audio coming from a radio transceiver on my sound card's microphone input. What i want to make is a simple software-based parrot repeater using Linux CLI tools like the sox suite and arecord. For it to work, i think a flow similar to the following must take place:
The audio that comes on the microphone subdevice is getting recorded in a buffer (file or RAM-based)
When the buffer stops filling (audio stopped), start playing it's content on the audio output device (it is connected to the radio's microphone input)
When it's over, empty the buffer and start expecting step 1 to occur again
I'm looking for an elegant way to implement the logic behind step 2. Is there a CLI tool that i can use for that, so i can pipe the microphone audio taken with arecord to it and play the output of the buffer with sox?
Try looking at this. I did this on a raspberry pi a little while ago, only I made a voice changer.
https://www.instructables.com/Halloween-Voice-Changer-With-Raspberry-Pi/
Basically, play "|rec --buffer 2048 -d" takes recorded sound and puts it in a buffer that is passed in 4096 bit (byte?) chunks to play. -d stands for duration, and if left blank defaults to 0, and will run until killed. If you want to play with the options, there is some helpful info in the links.
Good luck with your project!
We have a setup with a Windows 7 machine where we installed Dante Virtual Soundcard and start that soundcard with ASIO capabilities. The soundcard will receive audio over the network from a Tesira server. We want to capture the audio to files (highly preferring each channel to a separate file). The files will be played back on a later moment. There will likely be 6 channels or more.
In the same setup we use ffmpeg to capture some video which is working fine, with Direct Show. So for audio we wanted to use the same setup, since ffmpeg is able to record audio as well. However, there seems to be no option to select the ASIO devices which the virtual soundcard probably creates. So the question is what command line to use for ffmpeg, or what to install? Or which other program can record ASIO from command line?
I already tried installing:
Asio4all (actually wrong way around)
sox (don't know why actually)
HiFi Cable Asio Bridge (from VB-audio, not enough channels even with donate version)
Voicemeeter (from VB-Audio, not enough channels and actually mixes down)
O Deus Asio link, this might be an interesting option but it did not let me configure any route, any suggestions?
One thing I noticed is that the virtual soundcard can also be set to use WDM. Then I can see the devices with ffmpeg -list_devices true -f dshow -i duymmy, but recording does not yield any result, I have to ctrl-c to make it stop instead of q, and the file is zero bytes. Supposedly this is because the data over the network is all ASIO formatted and the Tesira Server cannot send "WDM data". FFmpeg stops at selecting the capture pin for audio only
EDIT:
I ran ffmpeg with high verbosity and when selecting the WDM soundcard it stops at Selecting pin Capture on audio only. Also when requesting the options it gives the same line for 22 times: min ch=1 bits=8 rate= 11025 max ch=2 bits=16 rate= 44100
You might use Voicemeeter instead of HIFI-Cable / ASIO-Bridge. Voicemeeter is a virtual audio device mixer able to connect everything together, any audio point, in any interface and any app together (including ASIO DAW)... Download & User Manual on www.voicemeeter.com
To answer my own question: it is not possible to capture sound from an ASIO device with ffmpeg. Maybe I will write the code for it if I need it...
I could however solve my issues by separating the two streams of audio data we have (AVB and Dante). These where on the same switch and maybe it is a bug in the firmware, maybe misconfiguration.
Thanks for your help!
How do I get the output from an ASIO device to IceCast2 or FFMpeg?
Duplicate?
And if not, Place the output for ffmpeg -f dshow -i "audio=your_device_name_in_dshow" -list_options