I am working on a project that involves using a lot of found audio clips (some new, some very old archival and poor quality etc).
I am trying to figure out a way to have all audio clips to be of a similar quality (if this is possible) and play at a similar volume?
I have use of both audacity and ableton...any suggestions would be great.
What you are asking for is commonly called normalization. There are several tools that can do it, including commandline tools and also audacity.
You'll find the tool in audacity under Effect > Normalize...
You can select multiple audio tracks.
You could also consider using a limiter and/or a compressor on your track. Have a look in the Live effect reference for more info on these: https://www.ableton.com/en/manual/live-audio-effect-reference/
The results will not be as good as applying normalization by hand, but it will be a lot quicker.
Related
I recently wondered if it is possible to zoom or move things in FFMPEG based on an audio source.
I already played around with complex filters as they allow some audio visualization but didn't really manage to move/zoom things based on sound. See good examples of complex filters used for audio visualization at: https://hhsprings.bitbucket.io/docs/programming/examples/ffmpeg/audio_visualization/index.html
My current situation is that i have multiple inputs which one of should react on sound/maybe even on special frequency's.
I'm currently researching an problem regarding DOA (direction of arrival) regression for an audio source, and need to generate training data in the form of audio signals of moving sound sources. In particular, I have the stationary sound files, and I need to simulate a source and microphone(s) with the distances between them changing to reflect movement.
Is there any software online that could potentially do the trick? I've looked into pyroomacoustics and VA as well as other potential libraries, but none of them seem to deal with moving audio sources, due to the difficulties in simulating the doppler effect.
If I were to write up my own simulation code for dealing with this, how difficult would it be? My use case would be an audio source and a microphone in some 2D landscape, both moving with their own velocities, where I would want to collect the recording from the microphone as an audio file.
Some speculation here on my part, as I have only dabbled with writing some aspects of what you are asking about and am not experienced with any particular libraries. Likelihood is good that something exists and will turn up.
That said, I wonder if it would be possible to use either the Unreal or Unity game engine. Both, as far as I can remember, grant the ability to load your own cues and support 3D including Doppler.
As far as writing your own, a lot depends on what you already know. With a single-point mike (as opposed to stereo) the pitch shifting involved is not that hard. There is a technique that involves stepping through the audio file's DSP data using linear interpolation for steps that lie in between the data points, which is considered to have sufficient fidelity for most purposes. Lot's of trig, too, to track the changes in velocity.
If we are dealing with stereo, though, it does get more complicated, depending on how far you want to go with it. The head masks high frequencies, so real time filtering would be needed. Also it would be good to implement delay to match the different arrival times at each ear. And if you start talking about pinnas, I'm way out of my league.
As of now it seems like Pyroomacoustics does not support moving sound sources. However, do check a possible workaround suggested by the developers here in Issue #105 - where the idea of using a time-varying convolution on a dense microphone array is suggested.
I am starting a project to test the audio performance on linux.
What I need to do is to play the audio on our websystem and check the audio quality (or just check it has audio output) on linux.
I am going to record the audio on linux with ffmpeg. Is there any other better choice?
I don't know how to (automation) check I recorded is what I played, as well as the quality of recorded audio.
I think what you need is PESQ (Perceptual Evaluation of Sound Quality). However I have not found anything which is open source/free and out of the box.
You can download the recommendation from here:
http://www.itu.int/rec/T-REC-P.862-200511-I!Amd2/en
Basically this is the reference implementation of PESQ.
Sevana has an audio quality analyser which is not an ITU standard, it is AQuA:
http://www.sevana.fi/aqua_wiki.php
It is available for linux but I think you have to pay for it.
You can also check the similarities for two audio files with cross-correlation, please refer to here:
https://dsp.stackexchange.com/questions/736/how-do-i-implement-cross-correlation-to-prove-two-audio-files-are-similar
I just learned that lot of people are using Matlab or Octave to generate the necessary data, for example:
http://bagustris.blogspot.ie/2011/11/calculate-time-lag-from-cross.html
I undertaking a personal project which involves the development of a system which will automatically generate audio thumbnail clips (about 30 seconds in length) from a full length track.
In order to do this I want to look at the energy and pitch of the audio to try and correctly identify its major structural features.
Is there any open source software available that can do energy/pitch extraction? If not I will start looking into alternative methods using MATLAB.
Thanks!
YAAFE (Yet Another Audio Feature Extractor) http://yaafe.sourceforge.net/ does audio feature extraction in MATLAB, Python and C.
You might want to look into the Echo Nest API. It has a lot of audio analysis capabilities, and I know there's a script bundled in the Remix package that can automagically turn songs into shorter or longer versions (I believe the script is called earworm).
Audacity may do it.
Try JAudio which can extract features from an audio.
MARSYAS contains bextract for analysis, can find MFCCs and various other timbral and spectral features. http://marsyas.info/
I am curious. How would one implement the most simple audio engine ever? I have something like a stream for audio data in mind using your default audio device. Playing a lot with RtAudio, I think if one could drop some of the features, this would be possible. Someone any idea where to start?
I would do it (did do it) like this:
http://ccan.ozlabs.org/info/wwviaudio.html
Well there is no reason why you can't create an audio engine that has a trivially simple interface:
audioEngine.PlayStream(myStream)
The audio engine would then periodically read data from that stream and send it to the soundcard. The reason audio engines tend to be more complicated than this, is that there are all kinds of parameters you might want to control, including latency of playback, sample rate, bit depth, as well as often the need to convert audio between formats. Add in the problems of repositioning streams, and synchronizing multiple streams, supporting multiple audio driver APIs etc, and soon you have an audio engine as complicated as any other.
Thank you for your answers.
to .Mark Heath:
yes of course I know that there might be a lot of parameters to tweak be it the filter cutoff, resonance, delay timing etc etc ..
I was just curious how to build an audio engine as simple as possible and modular as possible. The major intention I had in mind was to rebuild the gameboy soundchip ( again here, there a lot of implementations ie. JavaBoy).
to.smcameron
It seems that ccan/wwviaaudio has a dependency to libvorbis / portaudio (version >=19), that would yield the same effect as using rtaudio ( which is, compared to other realtime audio interface having build in asio support, rather small). However, I will give it a try.
regards,
audax