I have a program that generates a PCM Audio stream (as a MemoryStream). I want to send this to an audio input (recording) device in windows so that an application that is set to get input from that recording device will get the output generated by my program as input.
Any simple way to tackle this? If it is going to help, I can generate Waveform output stream from my program.
As an additional complication, any solution that depends on looping back the audio output to input wont work because I have another application that is already doing that to capture the default output.
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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 have an idea that I have been working on, but there are some technical details that I would love to understand before I proceed.
From what I understand, Linux communicates with the underlying hardware through the /dev/. I was messing around with my video cam input to zoom and I found someone explaining that I need to create a virtual device and mount it to the output of another program called v4loop.
My questions are
1- How does Zoom detect the webcams available for input. My /dev directory has 2 "files" called video (/dev/video0 and /dev/video1), yet zoom only detects one webcam. Is the webcam communication done through this video file or not? If yes, why does simply creating one doesn't affect Zoom input choices. If not, how does zoom detect the input and read the webcam feed?
2- can I create a virtual device and write a kernel module for it that feeds the input from a local file. I have written a lot of kernel modules, and I know they have a read, write, release methods. I want to parse the video whenever a read request from zoom is issued. How should the video be encoded? Is it an mp4 or a raw format or something else? How fast should I be sending input (in terms of kilobytes). I think it is a function of my webcam recording specs. If it is 1920x1080, and each pixel is 3 bytes (RGB), and it is recording at 20 fps, I can simply calculate how many bytes are generated per second, but how does Zoom expect the input to be Fed into it. Assuming that it is sending the strean in real time, then it should be reading input every few milliseconds. How do I get access to such information?
Thank you in advance. This is a learning experiment, I am just trying to do something fun that I am motivated to do, while learning more about Linux-hardware communication. I am still a beginner, so please go easy on me.
Apparently, there are two types of /dev/video* files. One for the metadata and the other is for the actual stream from the webcam. Creating a virtual device of the same type as the stream in the /dev directory did result in Zoom recognizing it as an independent webcam, even without creating its metadata file. I did finally achieve what I wanted, but I used OBS Studio virtual camera feature that was added after update 26.0.1, and it is working perfectly so far.
I am trying to do some audio debugging on my Linux system.
I learned how to record the sound of the current playing media but how can I get the PCM data without DAC/ADC?
I mean, just like wireshark or tcpdump tool, is there some sort of alsadump that I can make use of?
I want to do bit-exact comparison of the output PCM data to make sure the audio processing algorithm (which is an executable binary) worked correctly.
Thanks a lot.
I have a windows phone 8 app which plays audio streams from a remote location or local files using the BackgroundAudioPlayer. I now want to be able to add audio effects, for example, reverb or echo, etc...
Please could you advise me on how to do this? I haven't been able to find a way of hooking extra audio processing code into the pipeline of audio processing even through I've read much about WASAPI, XAudio2 and looked at many code examples.
Note that the app is written in C# but, from my previous experience with writing audio processing code, I know that I should be writing the audio code in native C++. Roughly speaking, I need to find a point at which there is an audio buffer containing raw PCM data which I can use as an input for my audio processing code which will then write either back to the same buffer or to another buffer which is read by the next stage of audio processing. There need to be ways of synchronizing what happens in my code with the rest of the phone's audio processing mechanisms and, of course, the process needs to be very fast so as not to cause audio glitches. Or something like that; I'm used to how VST works, not how such things might work in the Windows Phone world.
Looking forward to seeing what you suggest...
Kind regards,
Matt Daley
I need to find a point at which there is an audio buffer containing
raw PCM data
AFAIK there's no such point. This MSDN page hints that audio/video decoding is performed not by the OS, but by the Qualcomm chip itself.
You can use something like Mp3Sharp for decoding. This way the mp3 will be decoded on the CPU by your managed code, you can interfere / process however you like, then feed the PCM into the media stream source. Main downside - battery life: the hardware-provided codecs should be much more power-efficient.
I need a program that I can pipe a raw PCM stream into, and will send wav files to stdout.
Since the input is a stream, I can't just add headers to it, but I can for every x seconds (For example, program reads ten seconds of the stream, and outputs a ten second long wav file, reads the next ten seconds, outputs a wav file, and so on).
Is there a program that can do this? It needs to run on Linux.
Check out sox, the Swiss Army knife of sound processing.
This is a dead simple program to write, in fact, if you can guarantee exactly the same number of bytes of PCM data in each wave file, you can just pre-create a standard wave file header for that amount of PCM data and then just cat wavehdr + chunk-of-streamdata
mplayer is quite good for this kind of tasks. It comes with two interface, mencoder and mplayer. mencoder as all the command line switch you want to transform raw pcm into various sound format.
It also come with a very long man page that explains all the options quite well.