I want to make an application that counts the speaking time of each speaker in an audio recording. I don't care about doing full voice recognition and transcribing every word in the recording, I just want the speaking time of each voice.
Is there a piece of software that provides such feature?
If possible, I would like to avoid using a third-party service (such as Google Cloud) to achieve this, and I would like the solution to be light enough to run on a modern smartphone.
Thank you for your help.
I had the same idea. Check this out https://github.com/pyannote/pyannote-audio
Haven't tried it myself yet. Will add an edit after.
Related
I have recorded an audio.
I dont know how it happened that only one sided speech is recorded and the other speech is recorded with a very low sound.
Is there any solution to amplify the other side signal.
any help would be much appreciated.
This question is probably more appropriately asked at a forum where recording and mixing is discussed. For example: https://sound.stackexchange.com/
The ideal would be to improve your recording situation, to control factors so the sound are more closely matched. (Match microphones, isolate the speakers from environmental sounds, optimize input levels, etc.)
After that, the next option or step is to pre-process your audio files with a tool like Audacity. Use this or another DAW (Digital Audio Workstation) tool to match amplitudes or employ noise filtering or a range of other tools.
Audio processing is both tricky (an "art") and cpu intensive, so it's good to get as much of this handled as possible before the sounds are imported into a program.
Here is what i like to achieve:
I like to play around in creating "new" software / hardware instruments.
Sound processing and creation is always managed by software. But one could play the instrument via ultrasonic distance sensor for example. Another idea is to start playback when someone interrupts the light of a photoelectric barrier and so on....
So the instrument would play common sounds, but has to be used in an unusal way. For example, the ultrasonic instrument would play a sound if it detects something in a certain distance. The sound could be manipiulated in pitch for example if the distance gets smaller.
Basically i like to playback a sound sample and manipualte this in realtime.
I guess i have to use WAV samples for this, right? And which programming language do you think fits best for this task?
Edited after kevins hint: please kick me into the right direction - give me a hint where to start.
Thanks in advance
Since you're using the the Processing tag, you can try Processing.
It comes with a sound library like Minim or you can install beads which is great. There's actually a nice book on it: Sonifying Processing
You might find SuperColider fun as well.
The main thing is what are you comfortable with at the moment ?
If Processing syntax looks intimidating, you can actually try a different programming paradigm like data flow. In which case you can use PureData(free, opensource) or MaxMSP(very similar, but commercial). The idea is rather than typing instructions, you connect boxes with wires which is fun and the examples are great too.
If you're into c++ there are plenty of libraries. On the creative side, there's a nice set of libraries called OpenFrameworks that's easy and fun to use. If this is your cup of tea, have a peek at Maximilian.
Bottomline is: there are multiple options to achieve the same task. Choose the best tool for your (based on your background) or try each and see what you like best.
You asked "And which programming language do you think fits best for this task?" - I would also suggest using Processing. I have been used Processing to work with sounds previously. And in all cases I used Minim. It has many UgenS to generate sounds programmatically.
Also, you wants to integrate with some sensors. I'm not sure what types of sensors you will use, but Processing goes pretty well with different Arduino modules and sensors. Check this link for more direction.
Furthermore, you can export your project as .exe or executable .jar files. And their JS version (P5.js) works almost the same as the Java version.
I need to develop a program that toggles a particular audio track on or off when it recognizes a parrot scream or screech. The software would need to recognize a particular range of sounds and allow some variations in the range (as a parrot likely won't replicate its sreeches EXACTLY each time).
Example: Bird screeches, no audio. Bird stops screeching for five seconds, audio track praising the bird plays. Regular chattering needs to be ignored completely, as it is not to be discouraged.
I've heard of java libraries that have speech recognition with dictionaries built in, but the software would need to be taught the particular sounds that my particular parrot makes - not words or any random bird sound. In addition as I mentioned above, it would need to allow for slight variation in the sound, as the screech will likely never be 100% identical to the recorded version.
What would be the best way to go about this/what language should I look into?
Edit: Alternatively (and perhaps this would be a more simple solution), is there a way to make the audio toggle based on the volume of input? So it wouldn't matter what kind of sound the parrot makes, just how loud it is?
This question seems to be tightly related to voice recognition. I would recomend taking a look at this post: How to convert human voice into digital format?
I'm trying to develop an online application where the user writes some text and the software sings it back to the user.
I can currently generate the audio file with the words spoken by the computer using espeak, but I have no idea how to make it sound like a song, how to add rhythm to it.
I'm able to change the pitch and tempo using rubberband, but that's as far as I've gotten.
Does anyone have a clue how to make this happen?
If you want to use rubberband to change duration and pitch, then I think the hard part is going to be mapping from phonemes/syllables in the text to corresponding audio ranges in the speech systhesis output, for which I have no simple suggestion. (Ideally you'd get inside the speech synthesiser so that it would provide you with the mapping from phonemes to audio location.)
A simpler alternative might be to try Speech Synthesizer Markup Language - SSML. It has a "pitch" and "duration" elements that can absolutely specify pitch in Hz and duration in seconds. You can also specify volume, for controlling dynamics.
Given this, you could try to convert the text into a SSML document, and mark up words/syllables/phonemees with pitch/duration and volume attributes.
I've ended up using Festival's singing mode. It sounds reasonably well, except for the fact it only works with English voices.
I want to make a program that takes recorded speech and transforms it so it sounds like it's coming from a Texas TI-99. Do you have any good ideas and resources for how to go about that?
Most of those old speech synthesizers were build directly in-chip. Perhaps you could find a synthesizer that sounds like the chip, but if you really want the original sound, you would either have to simulate the chip (I don't know if it's a simple matter, perhaps the chip internals aren't published).
I only know because I burnt out a number of the Radio Shack speech synthesizer ICs before I managed to get a SP0256-AL2 working.
If you're more of a do-it yourself type guy, you need to find out which IC actually drove the speech synthesis in a TI-99, and then build the chip up on a bread board. That's what I was trying to do back then, and I managed to get the chip to speak, but lost patience after I fried my third chip due to a mis-wiring issue when I attempted to attach it to my PC's parallel port. I think this was the book I was using back then, but there's no cover art featured so it's hard to know for sure.
If you are familiar with how to use ROM images, there seems to be a gentleman that has managed to refeverse engineer the ROM image out of a SP0256-AL2. Look here for the image and the incredible granted permission to do the work and distribute the results.
You could start with open source that does something similar: Adding Robotic/Vocoder effect to your song using Audacity