audio | declarative use of Web Audio API | REST library

 by   ng-web-apis TypeScript Version: Current License: MIT

kandi X-RAY | audio Summary

kandi X-RAY | audio Summary

audio is a TypeScript library typically used in Web Services, REST, Nodejs applications. audio has no bugs, it has no vulnerabilities, it has a Permissive License and it has low support. You can download it from GitHub.

This library has AudioBufferService with fetch method, returning Promise which allows you to easily turn your hosted audio file into AudioBuffer through GET requests. Result is stored in service's cache so same file is not requested again while application is running.
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            kandi-support Support

              audio has a low active ecosystem.
              It has 84 star(s) with 2 fork(s). There are 5 watchers for this library.
              OutlinedDot
              It had no major release in the last 6 months.
              There are 0 open issues and 3 have been closed. On average issues are closed in 39 days. There are 5 open pull requests and 0 closed requests.
              It has a neutral sentiment in the developer community.
              The latest version of audio is current.

            kandi-Quality Quality

              audio has 0 bugs and 0 code smells.

            kandi-Security Security

              audio has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.
              audio code analysis shows 0 unresolved vulnerabilities.
              There are 0 security hotspots that need review.

            kandi-License License

              audio is licensed under the MIT License. This license is Permissive.
              Permissive licenses have the least restrictions, and you can use them in most projects.

            kandi-Reuse Reuse

              audio releases are not available. You will need to build from source code and install.
              Installation instructions are not available. Examples and code snippets are available.

            Top functions reviewed by kandi - BETA

            kandi's functional review helps you automatically verify the functionalities of the libraries and avoid rework.
            Currently covering the most popular Java, JavaScript and Python libraries. See a Sample of audio
            Get all kandi verified functions for this library.

            audio Key Features

            No Key Features are available at this moment for audio.

            audio Examples and Code Snippets

            Calculates a summary of the audio summary .
            pythondot img1Lines of Code : 54dot img1License : Non-SPDX (Apache License 2.0)
            copy iconCopy
            def audio_summary(tag,
                              tensor,
                              sample_rate,
                              max_outputs=3,
                              collections=None,
                              name=None):
              # pylint: disable=line-too-long
              """Outputs a `Summary` protocol buf  
            Generate the graph for the given audio data .
            pythondot img2Lines of Code : 26dot img2License : Non-SPDX (Apache License 2.0)
            copy iconCopy
            def run_graph(wav_dir, labels, input_layer_name, output_layer_name,
                          num_top_predictions):
              """Runs the audio data through the graph and prints predictions."""
              with tf.compat.v1.Session() as sess:
                # Feed the audio data as input to  
            Runs the audio loop .
            javadot img3Lines of Code : 20dot img3License : Permissive (MIT License)
            copy iconCopy
            @Override
                public void run() {
                    duration = 0;
            
                    try (final ByteArrayOutputStream out = new ByteArrayOutputStream(); final TargetDataLine line = getTargetDataLineForRecord();) {
            
                        int frameSizeInBytes = format.getFrameSize(  

            Community Discussions

            QUESTION

            Vuejs Webpack Compression Plugin not compressing
            Asked 2022-Mar-28 at 12:53

            I need help debugging Webpack's Compression Plugin.

            SUMMARY OF PROBLEM

            • Goal is to enable asset compression and reduce my app's bundle size. Using the Brotli algorithm as the default, and gzip as a fallback for unsupported browsers.
            • I expected a content-encoding field within an asset's Response Headers. Instead, they're loaded without the field. I used the Chrome dev tools' network tab to confirm this. For context, see the following snippet:
            • No errors show in my browser or IDE when running locally.

            WHAT I TRIED

            • Using different implementations for the compression plugin. See below list of approaches:
              1. (With Webpack Chain API)
            ...

            ANSWER

            Answered 2021-Sep-30 at 14:59

            It's not clear which server is serving up these assets. If it's Express, looking at the screenshot with the header X-Powered-By, https://github.com/expressjs/compression/issues/71 shows that Brotli support hasn't been added to Express yet.

            There might be a way to just specify the header for content-encoding manually though.

            Source https://stackoverflow.com/questions/69288974

            QUESTION

            Different behavior between match and unwrap
            Asked 2022-Mar-15 at 18:39

            I have done a small program, and it shows a weird behavior that I cannot explain. I am using rodio crate to try out some audio stuff.

            I have done two programs that, in my opinion, should give the same result.

            The first one I use matches to handle errors:

            ...

            ANSWER

            Answered 2021-Sep-30 at 19:08

            The issue is one of scoping and an implementation detail of rodio: the one critical item here is OutputStream::try_default(), it doesn't really matter how you handle Sink::try_new(&handle) it'll always behave the same, not so try_default, if you match or if let it it'll work fine, if you unwrap it it'll fail.

            But why would that be, the two should be equivalent. The answer is in the details of rodio, specifically of OutputStreamHandle:

            Source https://stackoverflow.com/questions/69393226

            QUESTION

            IndexError: tuple index out of range when I try to create an executable from a python script using auto-py-to-exe
            Asked 2022-Feb-24 at 15:03

            I have been trying out an open-sourced personal AI assistant script. The script works fine but I want to create an executable so that I can gift the executable to one of my friends. However, when I try to create the executable using the auto-py-to-exe, it states the below error:

            ...

            ANSWER

            Answered 2021-Nov-05 at 02:20
            42681 INFO: PyInstaller: 4.6
            42690 INFO: Python: 3.10.0
            

            Source https://stackoverflow.com/questions/69776492

            QUESTION

            Javascript: frame precise video stop
            Asked 2022-Jan-28 at 14:55

            I would like to be able to robustly stop a video when the video arrives on some specified frames in order to do oral presentations based on videos made with Blender, Manim...

            I'm aware of this question, but the problem is that the video does not stops exactly at the good frame. Sometimes it continues forward for one frame and when I force it to come back to the initial frame we see the video going backward, which is weird. Even worse, if the next frame is completely different (different background...) this will be very visible.

            To illustrate my issues, I created a demo project here (just click "next" and see that when the video stops, sometimes it goes backward). The full code is here.

            The important part of the code I'm using is:

            ...

            ANSWER

            Answered 2022-Jan-21 at 19:18

            The video has frame rate of 25fps, and not 24fps:

            After putting the correct value it works ok: demo
            The VideoFrame api heavily relies on FPS provided by you. You can find FPS of your videos offline and send as metadata along with stop frames from server.

            The site videoplayer.handmadeproductions.de uses window.requestAnimationFrame() to get the callback.

            There is a new better alternative to requestAnimationFrame. The requestVideoFrameCallback(), allows us to do per-video-frame operations on video.
            The same functionality, you domed in OP, can be achieved like this:

            Source https://stackoverflow.com/questions/70613008

            QUESTION

            How to detect if a device can play Dolby Atmos Sound with Javascript
            Asked 2021-Dec-24 at 06:57

            I am looking for a way to detect if the device I am using can support Dolby Atmos sounds.

            After searching around I found this call.

            https://github.com/w3c/media-capabilities/blob/main/explainer.md#spatial-audio

            ...

            ANSWER

            Answered 2021-Dec-24 at 06:57

            Detecting the codec doesn't necessarily detect whether the system can support Dolby Atmos

            Correct.

            What reliable way is there to detect if your system will truly support Dolby Atmos whether its with a receiver or a Dolby Atmos compliant sound bar.

            Unfortunately, this undetectable from the browser.

            The browser itself and even the OS doesn't always know what is downstream. Sorry for the bad news!

            Source https://stackoverflow.com/questions/70263091

            QUESTION

            FFMPEG's xstack command results in out of sync sound, is it possible to mix the audio in a single encoding?
            Asked 2021-Dec-16 at 21:11

            I wrote a python script that generates a xstack complex filter command. The video inputs is a mixture of several formats described here:

            I have 2 commands generated, one for the xstack filter, and one for the audio mixing.

            Here is the stack command: (sorry the text doesn't wrap!)

            ...

            ANSWER

            Answered 2021-Dec-16 at 21:11

            I'm a bit confused as how FFMPEG handles diverse framerates

            It doesn't, which would cause a misalignment in your case. The vast majority of filters (any which deal with multiple sources and make use of frames, essentially), including the Concatenate filter require that be the sources have the same framerate.

            For the concat filter to work, the inputs have to be of the same frame dimensions (e.g., 1920⨉1080 pixels) and should have the same framerate.

            (emphasis added)

            The documentation also adds:

            Therefore, you may at least have to add a ​scale or ​scale2ref filter before concatenating videos. A handful of other attributes have to match as well, like the stream aspect ratio. Refer to the documentation of the filter for more info.

            You should convert your sources to the same framerate first.

            Source https://stackoverflow.com/questions/70020874

            QUESTION

            How can I use value_counts() only for certain values?
            Asked 2021-Nov-23 at 00:53

            I want to extract how many positive reviews by brand are in a dataset which includes reviews from thousands of products. I used this code and I got a table including percentaje of positive and non-positive reviews. How can I get only the percentage of positive reviews by brand? I only want the "True" results in positive_review. Thanks!

            ...

            ANSWER

            Answered 2021-Nov-22 at 17:40

            Using the following toy DataFrame as an example:

            Source https://stackoverflow.com/questions/70069983

            QUESTION

            Audio recording in Python with Pyaudio, error ||PaMacCore (AUHAL)|| ... msg=Audio Unit: cannot do in current context
            Asked 2021-Nov-09 at 04:30

            I am using pyaudio to record sounds on my Mac BigSur 11.6 (20G165). Specifically, I'm redirecting sound from an application to the input using BlackHole, which works fine.

            It usually works fine but, sometimes, I get this error in the Terminal:

            ||PaMacCore (AUHAL)|| Error on line 2500: err='-10863', msg=Audio Unit: cannot do in current context

            Any idea why or how I could prevent it from happening (like, waiting until PaMacCore is ready to record again or something)?

            I already tried reinstalling but it doesn't help

            ...

            ANSWER

            Answered 2021-Nov-03 at 16:29

            Apparently the problem were mismatched bitrates in BlackHole's aggregated output device. I was aggregating Blackhole's output (44,1kHz) and the Mac Speakers (48kHz). This did not cause any consistent bad behaviour but sometimes led to these errors.

            Source https://stackoverflow.com/questions/69571374

            QUESTION

            Java mp4parser to create a single .m4s fragment— invalid moov box
            Asked 2021-Oct-22 at 06:01

            This use case is a service that manually encodes a series of uncompressed .wav media segments into .m4s fragments for broadcast via MPEG-DASH, using ffmpeg to compress the .wav to .aac and sannies/mp4parser to assemble the aac audio into an .m4s media fragment.

            I created this public GitHub project to reproduce the issue in its entirety.

            For example, here's the custom CustomFragmentMp4Builder.java class.

            It's critical that we be able to designate this single .m4s fragment with a sequence number (index) that we will manually increment for each media segment.

            The objective is to build an .m4s fragment comprising the box types SegmentTypeBox, SegmentIndexBox, and MovieFragmentBox. As For reference, I have used mp4parser to inspect an .m4s fragment that was generated via ffmpeg -f hls. This specification is available here as a .yaml file

            My implementation creates an MP4 without error. But, when the unit test attempts to read the file that the ChunkMp4Builder just wrote to a temp folder:

            ...

            ANSWER

            Answered 2021-Oct-21 at 20:57

            Your m4s segments are invalid due to an incorrect mdat atom size.

            For example in test5-128k-151304042.m4s the mdat is marked as having a length of 16 bytes but there is data at the end and file size is 164884.

            The parser then attempts to read an invalid offset. avc5 is not an atom but actually part of the string "Lavc58.54.100". The length read as 3724673100 is also invalid and greater than the max for a 32-bit integer, hence the invalid cast to int.

            In your implementation you have:

            Source https://stackoverflow.com/questions/69625970

            QUESTION

            Node.js - Client to Server WebRTC with P2P
            Asked 2021-Aug-26 at 04:13

            After looking to implement WebRTC with a Client to Server model (like Discord), I came to the conclusion that the way to do this is to have 2 clients - the server and client. Audio streams can be overlayed and sent back to the user in 1 single stream.

            backend/server.js

            ...

            ANSWER

            Answered 2021-Aug-26 at 04:13

            A possible solution can be: Create a MediaRecorder object, which can record the media streams on the client-side. This object emits data chunks over time. You can send these chunks via WebSocket to the server. On the server side, you can do what you want with the data chunks. For more details, you can check this https://mux.com/blog/the-state-of-going-live-from-a-browser/.

            https://developer.mozilla.org/en-US/docs/Web/API/MediaRecorder Another solution can be: Making a node.js application a PEER with WebRTC

            Source https://stackoverflow.com/questions/68821533

            Community Discussions, Code Snippets contain sources that include Stack Exchange Network

            Vulnerabilities

            No vulnerabilities reported

            Install audio

            You can download it from GitHub.

            Support

            You can use following audio nodes through directives of the same name (prefixed with wa standing for Web API):.
            Find more information at:

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            CLONE
          • HTTPS

            https://github.com/ng-web-apis/audio.git

          • CLI

            gh repo clone ng-web-apis/audio

          • sshUrl

            git@github.com:ng-web-apis/audio.git

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