audiodata | JS interface to encapsulate the Audio Data API methods | Runtime Evironment library

 by   notmasteryet JavaScript Version: Current License: No License

kandi X-RAY | audiodata Summary

kandi X-RAY | audiodata Summary

audiodata is a JavaScript library typically used in Server, Runtime Evironment, Nodejs applications. audiodata has no bugs, it has no vulnerabilities and it has low support. You can download it from GitHub.

The Audio Data API Objects library. Source can be found at Start examples and wiki at The working examples can be found in examples/ folder. The testing requires JS shell from mozilla-central run in the testing/ folder (specify the JSSHELL variable with path to the binary): make run-tests. Also, jslint can be run (download jslint.js into the testing/ folder): make run-lint. To build the documentation, run in the docs/ folder: make build-docs.
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              audiodata has a low active ecosystem.
              It has 65 star(s) with 6 fork(s). There are 6 watchers for this library.
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              It had no major release in the last 6 months.
              There are 1 open issues and 1 have been closed. There are no pull requests.
              It has a neutral sentiment in the developer community.
              The latest version of audiodata is current.

            kandi-Quality Quality

              audiodata has no bugs reported.

            kandi-Security Security

              audiodata has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

            kandi-License License

              audiodata does not have a standard license declared.
              Check the repository for any license declaration and review the terms closely.
              OutlinedDot
              Without a license, all rights are reserved, and you cannot use the library in your applications.

            kandi-Reuse Reuse

              audiodata releases are not available. You will need to build from source code and install.

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            audiodata Key Features

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            audiodata Examples and Code Snippets

            No Code Snippets are available at this moment for audiodata.

            Community Discussions

            QUESTION

            async / await function and return value
            Asked 2021-Jun-06 at 01:54

            From the highest level, I'm trying to pass a Blob to a function that will transcribe the data and return the transcript. I'm struggling to get the async parts of the process lined up correctly. Any insight would be appreciated.

            The two files I'm working with are below. In the record.jsx file, I'm calling the googleTranscribe function ( that is located in the second file ) to do the transcription work and hopefully return the transcript. This is where I'm running into the problem - I can get the transcript but cannot return it as a value. I know I'm doing something wrong with async / await / promises, I just can't quite figure out what I'm doing wrong.

            record.jsx

            ...

            ANSWER

            Answered 2021-Jun-06 at 01:48

            The problem is in the second file. The line with your Axios should be modified as such:

            Source https://stackoverflow.com/questions/67855162

            QUESTION

            WAVE file unexpected behaviour
            Asked 2021-Jun-04 at 09:08

            I am currently trying to make a .wav file that will play sos in morse.

            The way I went about this is: I have a byte array that contains one wave of a beep. I then repeated that until I had the desired length. After that I inserted those bytes into a new array and put bytes containing 00 (in hexadecimal) to separate the beeps.

            If I add 1 beep to a WAVE file, it creates the file correctly (i.e. I get a beep of the desired length). Here is a picture of the waves zoomed in (I opened the file in Audacity): And here is a picture of the entire wave part:

            The problem now is that when I add a second beep, the second one becomes completely distorted: So this is what the entire file looks like now:

            If I add another beep, it will be the correct beep again, If I add yet another beep it's going to be distorted again, etc. So basically, every other wave is distorted.

            Does anyone know why this happens?

            Here is a link to a .txt file I generated containing the the audio data of the wave file I created: byteTest19.txt

            And here is a lint to a .txt file that I generated using file format.info that is a hexadecimal representation of the bytes in the .wav file I generated containing 5 beeps (with two of them, the even beeps being distorted): test3.txt

            You can tell when a new beep starts because it is preceded by a lot of 00's.

            As far as I can see, the bytes of the second beep does not differ from the first one, which is why I am asking this question.

            If anyone knows why this happens, please help me. If you need more information, don't hesitate to ask. I hope I explained well what I'm doing, if not, that's my bad.

            EDIT Here is my code:

            ...

            ANSWER

            Answered 2021-Jun-04 at 09:07

            The problem

            Your .wav file is Signed 16 bit Little Endian, Rate 44100 Hz, Mono - which means that each sample in the file is 2 bytes long, and describes a signed amplitude. So you can copy-and-paste chunks of samples without any problems, as long as their lengths are divisible by 2 (your block size). Your silences are likely of odd length, so that the 1st sample after a silence is interpreted as

            Source https://stackoverflow.com/questions/67827578

            QUESTION

            redux state takes empty object by dispatching Blob
            Asked 2021-Jun-01 at 14:20

            I am currently working on a project in which I should record user's voice and then send it to server. This project is being done in ReactJS and Redux. I can record the voice correctly but when I dispatch this blob with the aim of updating the store, it takes empty object.

            My Recorder component

            ...

            ANSWER

            Answered 2021-Jun-01 at 14:20

            You should not put non-serializable values into the Redux store, and a Blob is most definitely a non-serializable value.

            It's possible that the actual data is there correctly, but the DevTools is unable to display that value at all because A) it's a Blob and not anything readable, and B) that value can't be serialized properly for display in the DevTools.

            This is actually a good example of why we tell users not to do this :)

            Source https://stackoverflow.com/questions/67785620

            QUESTION

            get transform component via string to track player
            Asked 2021-May-11 at 16:14

            im trying to grab the transform component via t.find and then via the transform component, grab the game object component, and then t.LookAt can work, and look at the player.

            this is the code:

            ...

            ANSWER

            Answered 2021-May-09 at 19:36

            Here are a few issues with the updated snippet, not sure if they will fix all of the errors you are getting.

            You are re-using a variable name t. I would also refrain from naming any global variables very short random letters as it can get confusing. What is this a Transform of? If it is the transform of a player, possibly name it PlayerTransfom.

            As I mentioned you are re-using a variable name which is not allowed especially with different types as well as re-initilization. It is fine to re-assign a variable such that it is updated, manipulating, changing, etc. the value, but re-declaring the value is not allowed in c# and I do not believe allowed in any other language (at least that I know of).

            To be specific, the line target t = hit.transform.GetComponent(); is delcaring t as the script component type target, but you have already declared a Transform named t above it as public Transform t;.

            Another issue is you are attempting to assign a variable using a non-static method in a global setting. The line specifically is public Transform player = Transform.Find(playername);. You are not able to set the transform like this outside of a method as the Find method is not static. Change it to be.

            Source https://stackoverflow.com/questions/67451104

            QUESTION

            Oboe stops rendering audio if there is too many data to be rendered
            Asked 2021-May-06 at 10:52

            I'm trying to implement Oboe library into my application, so I can perform low latency audio playing. I could perform panning, playback manipulation, sound scaling, etc. I've been asking few questions about this topic because I'm completely new to audio worlds.

            Now I can perform basic things which internal Android audio class such as SoundPool provides. I can play multiple sounds simultaneously without noticeable delays.

            But now I encountered another problem. So I made very simple application for example; There is button in screen and if user taps this screen, it plays simple piano sound. However fast user taps this button, it must be able to mix those same piano sounds just like what SoundPool does.

            My codes can do this very well, until I taps button too much times, so there are many audio queues to be mixed.

            ...

            ANSWER

            Answered 2021-May-06 at 10:52

            Does Oboe stop rendering audio if it takes too much time to render audio like situation above?

            Yes. If you block onAudioReady for longer than the time represented by numFrames you will get an audio glitch. I bet if you ran systrace.py --time=5 -o trace.html -a your.app.packagename audio sched freq you'd see that you're spending too much time inside that method.

            Did I reach limit of rendering audio, meaning that only limiting number of queues is the solution? or are there any ways to reach better performance?

            Looks like it. The problem is you're trying to do too much work inside the audio callback. Things I'd try immediately:

            • Different compiler optimisation: try -O2, -O3 and -Ofast
            • Profile the code inside the callback - identify where you're spending most time.
            • It looks like you're making a lot of calls to sin and cos. There may be faster versions of these functions.

            I spoke about some of these debugging/optimisation techniques in this talk

            One other quick tip. Try to avoid raw pointers unless you really have no choice. For example AudioStream* stream; would be better as std::shared_ptr and std::vector* players; can be refactored to std::vector players;

            Source https://stackoverflow.com/questions/67415335

            QUESTION

            SwiftUI: AVAudioPlayer does not stop
            Asked 2021-May-05 at 23:35

            I have an AVAudioPlayer with some key data in an Observable Object:

            ...

            ANSWER

            Answered 2021-May-05 at 23:35

            If you want to mutate a struct owned by a SwiftUI View, it should be be a @State variable -- trying to mutate non-@State variables will at-best not compile (the compiler should tell you that you Cannot use mutating member on immutable value: 'self' is immutable) and, at worst, (if you've somehow found a way around that compiler error), cause unexpected consequences.

            Future readers, see the comments on the original question for a more detailed explanation of how this conclusion was reached

            Source https://stackoverflow.com/questions/67409717

            QUESTION

            How to pan audio sample data naturally?
            Asked 2021-Apr-13 at 20:10

            I'm developing Flutter plugin which is targeting only Android for now. It's kind of synthesis thing; Users can load audio file into memory, and they can adjust pitch (not pitch shift) and play multiple sound with the least delay using audio library called Oboe.

            I managed to get PCM data from audio files which MediaCodec class supports, and also succeeded to handle pitch by manipulating playback via accessing PCM array manually too.

            This PCM array is stored as float array, ranging from -1.0 to 1.0. I now want to support panning feature, just like what internal Android class such as SoundPool. I'm planning to follow how SoundPool is handling panning. There are 2 values I have to pass to SoundPool when performing panning effect : left, and right. These 2 values are float, and must range from 0.0 to 1.0.

            For example, if I pass (1.0F, 0.0F), then users can hear sound only by left ear. (1.0F, 1.0F) will be normal (center). Panning wasn't problem... until I encountered handling stereo sounds. I know what to do to perform panning with stereo PCM data, but I don't know how to perform natural panning.

            If I try to shift all sound to left side, then right channel of sound must be played in left side. In opposite, if I try to shift all sound to right side, then left channel of sound must be played in right side. I also noticed that there is thing called Panning Rule, which means that sound must be a little bit louder when it's shifted to side (about +3dB). I tried to find a way to perform natural panning effect, but I really couldn't find algorithm or reference of it.

            Below is structure of float stereo PCM array, I actually didn't modify array when decoding audio files, so it should be common structure

            ...

            ANSWER

            Answered 2021-Apr-13 at 08:37

            Pan rules or pan laws are implemented a bit different from manufacturer to manufacturer.

            One implementation that is frequently used is that when sounds are panned fully to one side, that side is played at full volume, where as the other side is attenuated fully. if the sound is played at center, both sides are attenuated by roughly 3 decibels.

            to do this you can multiply the sound source by the calculated amplitude. e.g. (untested pseudo code)

            Source https://stackoverflow.com/questions/67062207

            QUESTION

            Record mic with pyaudio
            Asked 2021-Apr-12 at 12:34

            I try to record my mic with pyaudio. So I use the example program:

            ...

            ANSWER

            Answered 2021-Apr-12 at 12:34

            You can try to see whether you are using the right input device. Add the input_device_index={the right input device} argument to the audio.open.

            You can check the ids of your devices like so: How to select a specific input device with PyAudio

            Source https://stackoverflow.com/questions/67057922

            QUESTION

            "No filename found in "Content-Disposition" header. A file name in the format "filename=FILENAME" must be provided"
            Asked 2021-Mar-24 at 20:14

            i'm trying to upload audio file from ionic 4 to drupal 8 site using REST API code is :

            ...

            ANSWER

            Answered 2021-Mar-24 at 20:14

            the solutions is to change the header to

            Source https://stackoverflow.com/questions/66788103

            QUESTION

            Django and JavaScript: Using last uploaded file in an object list within in a Javascript file
            Asked 2021-Mar-21 at 07:42

            Edit: this question started as a question about copying files in Django but it turned out that the better way to achieve my aim of accessing files in JavaScript could be achieved directly.

            Original question

            I want to copy the latest uploaded mp3 file from the object list in my first model (which uses the default media folder) to a new folder called ‘latest’ and I also want to rename the new copy ‘latest.mp3’. This is so that I have a known filename to be able to process the latest uploaded file using Javascript. I wish to also keep the original, unaltered, uploaded file in the object list of my first model.

            The below is what I have so far but it doesn’t work: I don’t get any traceback error from the server or from the browser. However, the copy isn’t made in the ‘latest/’ folder. I believe I am doing more than one thing wrong and I am not sure if I should be using CreateView for the SoundFileCopy view. I am also not sure when the process in SoundFileCopy view is triggered: I am assuming it happens when I ask the browser to load the template ‘process.html’.

            I am using Django 3.1.7

            If anyone can help me by letting me know what I need to put in my models, views and template to get this to work I would be very grateful.

            Currently my models are:

            ...

            ANSWER

            Answered 2021-Mar-20 at 12:38

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