kandi X-RAY | atldotnet Summary
kandi X-RAY | atldotnet Summary
This library is aimed at giving .NET developers a managed, portable and easy-to-use library to read and write metadata from digital audio files and playlists with one single unified API, whatever the underlying format. You'll find more working code on the Code snippets section of the Documentation, including what you need to manage embedded pictures (e.g. cover), chapters , lyrics and playlists.
Top functions reviewed by kandi - BETA
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Trending Discussions on Audio
I'm making a speech-to-text tool. I'm capturing audio in real time (using Web audio api from Chrome) and sending it to a server to convert the audio to text.
I'd like to extract pieces of the whole audio cause I only want to send sentences, avoiding silences. (cause the api I use has a cost). The problem is that I don't know how to convert the whole audio into pieces.
I was using
MediaRecorder to capture the audio
ANSWERAnswered 2022-Mar-22 at 12:33
I've found the answer to my own question, I was using the wrong approach.
What I need to use to get the raw audio inputs and be able to manipulate them is the AudioWorkletProcessor.
This video helped me to understand the theory behind:
And this article helped me understand how to make use of it: https://developer.mozilla.org/en-US/docs/Web/API/Web_Audio_API/Using_AudioWorklet
I writing my code within a Jupyter notebook in VS Code. I am hoping to play some of the audio within my data set. However, when I execute the cell, the console reports no errors, produces the widget, but the widget displays 0:00 / 0:00 (see below), indicating there is no sound to play.
Below, I have listed two ways to reproduce the error.
ANSWERAnswered 2022-Mar-15 at 00:07
Apologies for the late reply! In the future, please tag the questions with activeloop so it's easier to sort through (or hit us up directly in community slack -> slack.activeloop.ai).
adding [:,0] to 9th line will help fixing display on Colab as Audio expects one-dimensional data
Hoping to have two buttons, one to "Pause" one to "Play" the MP3 audio file. Buttons show but do not pause/play the audio, Aswell as the audio does not play whatsoever. Here's my code I'm using. Thanks.
ANSWERAnswered 2022-Mar-09 at 21:52
I'm trying to download an mp3 file on the click of a button, it downloads, but the file it downloads is not correct, it's a WAY smaller file than the original one (25 bytes), while the original one is 10MB.
It's my first time working with downloading stuff so I'm pretty clueless.
Here's some code:
ANSWERAnswered 2022-Feb-24 at 18:15
You can try this. Here you have to provide audio file source instead of image source. I did not try this code, But i assure you to it should work!
I am trying to build a graphical audio spectrum analyzer on Linux. I run an FFT function on each buffer of PCM samples/frames fed to the audio hardware so I can see which frequencies are the most prevalent in the audio output. Everything works, except the results from the FFT function only allocate a few array elements (bins) to the lower and mid frequencies. I understand that audio is logarithmic, and the FFT works with linear data. But with so little allocation to low/mid frequencies, I'm not sure how I can separate things cleanly to show the frequency distribution graphically. I have tried with window sizes of 256 up to 1024 bytes, and while the larger windows give more resolution in the low/mid range, it's still not that much. I am also applying a Hann function to each chunk of data to smooth out the window boundaries.
For example, I test using a mono audio file that plays tones at 120, 440, 1000, 5000, 15000 and 20000 Hz. These should be somewhat evenly distributed throughout the spectrum when interpreting them logarithmically. However, since FFTW works linearly, with a 256 element or 1024 element array only about 10% of the return array actually holds values up to about 5 kHz. The remainder of the array from FFTW contains frequencies above 10-15 kHz.
Here's roughly the result I'm after:
But this is what I'm actually getting:
Again, I understand this is probably working as designed, but I still need a way to get more resolution in the bottom and mids so I can separate the frequencies better.
What can I do to make this work?...
ANSWERAnswered 2022-Feb-17 at 11:22
What you are seeing is indeed the expected outcome of an FFT (Fourier Transform). The logarithmic f-axis that you're expecting is achieved by the Constant-Q transform.
Now, the implementation of the Constant-Q transform is non-trivial. The Fourier Transform has become popular precisely because there is a fast implementation (the FFT). In practice, the constant-Q transform is often implemented by using an FFT, and combining multiple high-frequency bins. This discards resolution in the higher bins; it doesn't give you more resolution in the lower bins.
To get more frequency resolution in the lower bins of the FFT, just use a longer window. But if you also want to keep the time resolution, you'll have to use a hop size that's smaller than the window size. In other words, your FFT windows will overlap.
I'm trying to use the
tag, and I want to have as many tracks playing as I add. But in the end, the very first track plays in a circle. How can this be fixed?
ANSWERAnswered 2022-Feb-08 at 12:49
- Add a skip button
- when clicked, call a
skipfunction that will
- pause if there is something playing then
- clear the playing/paused class clist
- increment the current to the next track
- then play
I'm trying to create a sound using Fourier coefficients.
First of all please let me show how I got Fourier coefficients.
(1) I took a snapshot of a waveform from a microphone sound.
- Getting microphone: getUserMedia()
- Getting microphone sound: MediaStreamAudioSourceNode
- Getting waveform data: AnalyserNode.getByteTimeDomainData()
The data looks like the below: (I stringified Uint8Array, which is the return value of
getByteTimeDomainData(), and added
length property in order to change this object to Array later)
ANSWERAnswered 2022-Feb-04 at 23:39
In golang I have taken an array ARR1 which represents a time series ( could be audio or in my case an image ) where each element of this time domain array is a floating point value which represents the height of the raw audio curve as it wobbles ... I then fed this floating point array into a FFT call which returned a new array ARR2 by definition in the frequency domain where each element of this array is a single complex number where both the real and the imaginary parts are floating points ... when I then fed this array into an inverse FFT call ( IFFT ) it gave back a floating point array ARR3 in the time domain ... to a first approximation ARR3 matched ARR1 ... needless to say if I then took ARR3 and fed it into a FFT call its output ARR4 would match ARR2 ... essentially you have this time_domain_array --> FFT call -> frequency_domain_array --> InverseFFT call -> time_domain_array ... rinse N repeat
I know Web Audio API has a FFT call ... do not know whether it has an IFFT api call however if no IFFT ( inverse FFT ) you can write your own such function here is how ... iterate across ARR2 and for each element calculate the magnitude of this frequency ( each element of ARR2 represents one frequency and in the literature you will see ARR2 referred to as the frequency bins which simply means each element of the array holds one complex number and as you iterate across the array each successive element represents a distinct frequency starting from element 0 to store frequency 0 and each subsequent array element will represent a frequency defined by adding
incr_freq to the frequency of the prior array element )
Each index of ARR2 represents a frequency where element 0 is the DC bias which is the zero offset bias of your input ARR1 curve if its centered about the zero crossing point this value is zero normally element 0 can be ignored ... the difference in frequency between each element of ARR2 is a constant frequency increment which can be calculated using
I want to make sure the latency between my app and the bluetooth headphones is accounted for, but I have absolutely no idea how I can get this value. The closest thing I found was:
BluetoothLEPreferredConnectionParameters.ConnectionLatency which is only available on Windows 11... Otherwise there isn't much to go on.
Any help would be appreciated.
ANSWERAnswered 2022-Jan-18 at 09:08
It's very difficult to get the exact latency because it is affected by many parameters - but you're on the right track by guessing that the connection parameters are a factor of this equation. I don't have much knowledge on UWP, but I can give you the general parameters that affect the speed/latency, and then you can check their availability in the API or even contact Windows technical team to see if these are supported.
When you make a connection with a remote device, the following factors impact the speed/latency of the connection:-
- Connection Interval: this specifies the interval at which the packets are sent during a connection. The lower the value, the higher the speed. The minimum value as per the Bluetooth spec is 7.5ms.
- Slave Latency: this is the value you originally mentioned - it specifies the number of packets that can be missed before a connection is considered lost. A value of 0 means that you have the fastest most robust connection.
- Connection PHY: this is the modulation on which the packets are sent. If both devices support 2MPHY, then the connection should be quicker.
- Data Length/MTU Extension: these are two separate features but I am looping them together becuase the effect is the same - more bytes are sent per packet, which results in a higher throughput. The maximum value is 251 bytes per packet.
You can find more information about these parameters here:-
- A Practical Guide to BLE Throughput
- Maximizing BLE Throughput: Everything You Need to Know
- Bluetooth 5 Speed - How to Achieve Maximum Throughput
And below are some other links that might help you understand what is supported on UWP:-
I've got a simple wav header reader i found online a long time ago, i've gotten back round to using it but it seems to replace around 1200 samples towards the end of the data chunk with a single random repeated number, eg -126800. At the end of the sample is expected silence so the number should be zero.
Here is the simple program:...
ANSWERAnswered 2022-Jan-07 at 21:55
WAV is just a container for different audio sample formats.
You're making assumptions on a wav file that would have been OK on Windows 3.11 :) These don't hold in 2021.
Instead of rolling your own Wav file reader, simply use one of the available libraries. I personally have good experiences using
libsndfile, which has been around roughly forever, is very slim, can deal with all prevalent WAV file formats, and with a lot of other file formats as well, unless you disable that.
This looks like a windows program (one notices by the fact you're using very WIN32API style capital struct names – that's a bit oldschool); so, you can download libsndfile's installer from the github releases and directly use it in your visual studio (another blind guess).
I need to create a sort of like guitar tuner.. thats recognize the sound frequencies and determines in witch chord i am actually playing. Its similar to this guitar tuner that i found online: https://musicjungle.com.br/afinador-online But i cant figure it out how it works because of the webpack files..I want to make this tool app backendless.. Someone have a clue about how to do this only in the front end?
i founded some old pieces of code that doesnt work together.. i need fresh ideas...
ANSWERAnswered 2021-Sep-21 at 01:29
I suppose it'll depend how you're building your application. Hard to help without much detail around specs. Though, here are a few options for you.
There are a few stream options, for example;
Or if you're using React;
Or if you're wanting to go real basic with some vanilla JS;
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