5

Quick answer - No I haven't heard of any but I've heard about research being done on the field. The greatest theoretical problem with de-mixing is knowing what to subtract. When we mix, we add A and B which are both known signals. But when we try to do the opposite, we need to know in advance what B is. This would render the process useless in most cases ...


4

Audio To Midi - Free plugin by Mind The Pressure. Translates audio to midi signal directly so it can be hooked up with any virtual instrument (direct download) WIDI Audio to MIDI - Not free. Somewhat similar to the above, but it has some stability problems.


4

What you are asking is impossible to determine from a spectrogram. Spectrograms indicate frequencies present, not the quality of the audio. There may be loss of fine detail in one that would barely show up on a spectrogram at all or there could be noise and artifacts introduced that would make the spectrogram look more full. Spectrograms do not ...


3

I just had a look at Harmor and I'm assuming your question is "How can resynthesis engines be so accurate at RECONSTRUCTING SOUND despite the time/frequency resolution trade-offs inherent in Fourier analysis?" Well put simply, the time frequency decomposition, be at a wavelet transformation or short time Fourier transform is just another representation of ...


2

Here is a tentative answer based on the 192kHz 24-bit Japanese female speech sample: First I generate the test files from the sample file, sample02.wav: for br in 8 16 24 32 40 48 56 64 80; do for lq in 0 2 5 7 9; do printf -v f0 "out_b%02d_q%d" "$br" "$lq" lame -b "$br" -q "$lq" sample02.wav "${f0}.mp3" sox "${f0}.mp3" -r 96k "$...


2

Essentia is an Open-source library and tools for audio and music analysis, description and synthesis It includes some pre-built extractors that you can use.


2

In concurrence with other answers; generally "not really feasible" is the answer. I've had a play with iZotope RX5, a very interesting editor; draws the signal as a 2D frequency graph and has tools for selecting regions (including changing regions) to isolate certain sounds and amplify or subdue them, but in the context I was operating it proved impossible ...


2

There's actually one software company that claims to provide credible unmixing : audionamix. I am not affiliated with them, neither can I confirm their claims but there are demos on their website.


2

The upper one is definitey the better one. Spectrograms (like the ones you produced with Spek) are used to display and inspect tendencies over time. For example if some noisy signal is present all the time, it will show up as a straight horizontal line somewhere: You can determine some general properties, like that 20 Khz cut, but an average frequency ...


1

For dimensionality reduction you need features to start with. You can for example extract MFCC’s or some other low-level features such as MPEG-7 descriptors. Then you can visualise them using PCA. TBH for this task you might be better of using t-SNE or UMAP to project this high dimensional data while preserving local clusters. Lastly, just have a look at ...


1

Take a look at Praat which has spectral and pitch analysis built in. Here is a video explaining how this is done.


1

According to this spectral output you are not dealing with 48kHz sampled audio. For some reason the audio has been decimated and the actual sampling rate is 24kHz. Consequently the 'nyquist' frequency is 12kHz. One of the key things to remember about digital audio, is that during the process of sampling analogue audio the spectral content is mirrored ...


1

You can obtain very accurate audio spectrum analysis with a software analyzer. You must nevertheless be aware of the potential distortion brought by your acquisition chain (microphone + preamp) and take that into account in tour analysis.


1

Best audio quality is difficult to assess objectively. What you can say from the given audiograms is (as you mentioned) that N° 3 includes more high frequencies, which is usually considered better. About the N° 2 audiogram : there's usually no benefit in transcoding a lossy file into another lossy file codec because whatever information has been lost in ...


1

You would rather turn to a program like Matlab or its open source equivalent Octave. Open your mono-channel audio data as a vector (using wavread for instance), and turn each sample value to its square value. Then compute the mean every N samples, N depending on the smoothness/time-precision you need.


1

It depends on the quality of the recording/venue/PA. The basic technique you use here is echo cancellation like that used in mobile phones: you try subtracting a filtered version of the original signal from the composite signal until they are no longer correlated. You'll have much better likelihood for success if you record the total mix before going ...


1

Can you isolate the intended instrument or voice for each mic input? NO. There is no practical way of doing this. This is the "holy grail" of audio processing that people have been pursuing for decades without success. Can you predict the levels from different microphones in some attempt to "automate" the setup? NO. Especially in transient setups (...


1

EDIT: Find a vocal sample that has a defined note and is low enough to be pitched upwards. Take a 250ms chop from this vocal, preferably one in which there is a slight amount of pitch movement (vibrato). I recommend using a spectral view of the sample rather than a waveform view to make an informed decision about what 250ms slice to use. Pitch the sample ...


1

I would use a bubble pop noise like when you stick your finger in your mouth and pop it out... Then use a crinkled paper noise. They also have crow noises with each bubble pop. The crow noises have been pitch shifted a little as they progress to a slightly higher pitch each time.


1

Hmmm...tricky. I'd say something like a wet pair of jeans smashed against the floor together with squeezing a balloon, all with a lot of compressor.


1

It sounds like a combination of a short hit sound and a long water splash sound. Maybe a little paper wrinkling noise can be added.


1

Not entirely sure what you mean with the following phrase? a waveform stored in frequency domain representation? However, with FM the frequencies of the partials can be easily calculated as a function of the carriers frequency and the modulators frequency, such that C, C+M, C+2M, C+3M, etc. and C-M, C-2M, C-3M, etc. give the partials. Calculating the ...


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