audiostream | Stream and transcode your music library over HTTP | Audio Utils library

 by   nickdesaulniers JavaScript Version: 0.0.2 License: No License

kandi X-RAY | audiostream Summary

kandi X-RAY | audiostream Summary

audiostream is a JavaScript library typically used in Telecommunications, Media, Media, Entertainment, Audio, Audio Utils applications. audiostream has no bugs, it has no vulnerabilities and it has low support. You can install using 'npm i audiostream' or download it from GitHub, npm.

Stream and transcode your music library. The goal is to make software you want to use, for instance an itunes-like audio jukebox player experience in the browser. HTML5 audio is tricky because not all browser’s support all codecs/containers. This app uses ffmpeg, a free open source media transcoder, to support all browsers.
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            kandi-support Support

              audiostream has a low active ecosystem.
              It has 40 star(s) with 8 fork(s). There are 6 watchers for this library.
              OutlinedDot
              It had no major release in the last 12 months.
              There are 15 open issues and 22 have been closed. On average issues are closed in 22 days. There are no pull requests.
              It has a neutral sentiment in the developer community.
              The latest version of audiostream is 0.0.2

            kandi-Quality Quality

              audiostream has 0 bugs and 0 code smells.

            kandi-Security Security

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

            kandi-License License

              audiostream 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

              audiostream releases are not available. You will need to build from source code and install.
              Deployable package is available in npm.
              Installation instructions are available. Examples and code snippets are not available.
              audiostream saves you 87 person hours of effort in developing the same functionality from scratch.
              It has 223 lines of code, 0 functions and 9 files.
              It has low code complexity. Code complexity directly impacts maintainability of the code.

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            Currently covering the most popular Java, JavaScript and Python libraries. See a Sample of audiostream
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            audiostream Key Features

            No Key Features are available at this moment for audiostream.

            audiostream Examples and Code Snippets

            Send microphone audio recorder from browser to google speech to text - Javascript
            JavaScriptdot img1Lines of Code : 114dot img1License : Strong Copyleft (CC BY-SA 4.0)
            copy iconCopy
            const express = require('express');
            const multer = require('multer');
            const fs = require('fs');
            
            const upload = multer();
            
            const app = express();
            const port = 3000;
            
            app.use(express.static('./'));
            
            async function testGoogleTextToSpeech(aud
            How to accept permission for getUserMedia() in jasmine testing?
            JavaScriptdot img2Lines of Code : 11dot img2License : Strong Copyleft (CC BY-SA 4.0)
            copy iconCopy
            const audioStream = { ... }; // create a mock audio stream with the appropriate methods spied on
            let promise = Promise.resolve(audioStream);
            spyOn(navigator.mediaDevices, 'getUserMedia', promise);
            
              it('init()', as
            RTCMulticonnection initiator no camera
            Lines of Code : 18dot img3License : Strong Copyleft (CC BY-SA 4.0)
            copy iconCopy
            function SilentStream(videoColor) {
              const canvas = document.createElement('canvas');
              const ctx = canvas.getContext('2d', {alpha: false});
              ctx.fillStyle = videoColor || 'black';
              ctx.fillRect(0,0,canvas.width, canvas.height);
              
            Obtain MediaStream from input device
            JavaScriptdot img4Lines of Code : 39dot img4License : Strong Copyleft (CC BY-SA 4.0)
            copy iconCopy
            // using async for brevity
            async function doit() {
              // first request both mic and camera
              const gUMStream = await navigator.mediaDevices.getUserMedia({video: true, audio: true});
              // create a new MediaStream with only the audioTrack

            Community Discussions

            QUESTION

            stream.addTrack working in Firefox but not Chrome
            Asked 2021-May-24 at 03:00

            I'm having an issue with a project i'm working on for my brother's company and wanted to see if anybody had any ideas on how to fix it.

            The company puts on virtual conferences for corporate events, nonprofits, etc. and its common with these things for people to need to pre-record their presentations with a powerpoint slideshow. For non-tech savvy clients (which is most of their clients), the only way to make this happen is to hire a contractor to work with with the client 1 on 1 to get their video recorded. My goal is to make a site where clients can easily record the video and slideshow themself, which would reduce their costs significantly.

            Its still very much a work in progress, but what i have so far is live here: https://ezav-redesign-hf4d3.ondigitalocean.app/

            The issue is, while the recording function works really well on Firefox, on Chrome it always ends up with no audio. You can see the full source code at the live site, but here is what I think should be the relevant part:

            ...

            ANSWER

            Answered 2021-May-24 at 02:42

            This is a known bug/limitation in Chrome, where they don't pass the audio stream of muted MediaElements to the graph at all anymore.

            Luckily in your case it seems you absolutely don't need to go through the MediaElement since you have access to the raw MediaStream. So all you have to do is to get rid of the AudioContext part entirely and just do

            Source https://stackoverflow.com/questions/67665995

            QUESTION

            How to record screen+audio+microphone with RecordRTC
            Asked 2021-May-24 at 02:11

            I am working on a project where I need the user to be able to record screen, audio and microphone. At the moment I could only make it recognize screen and audio.

            First I am capturing the screen and the audio from it and saving it to a variable. And then I am capturing that variable to show the a video component.

            ...

            ANSWER

            Answered 2021-May-24 at 02:11

            I fixed it by increasing a function where I capture the audio from the microphone

            Source https://stackoverflow.com/questions/67546247

            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

            createReceiver isn't a function
            Asked 2021-Apr-23 at 22:29

            I need some help. The bot joins the voice channel but then crashes when I speak into the voice channel. I am trying to receive voice packets then tell the console that I received packets. The error that I get when speaking is:

            ...

            ANSWER

            Answered 2021-Apr-23 at 22:29

            In discord.js v12, VoiceConnection#createReciever() was removed in favor of VoiceConnection#reciever. I imagine this is because they wanted to limit you to one receiver instead of letting you create however many you want, but I'm not certain. I'm sure you can find a reason in the release notes.

            Source https://stackoverflow.com/questions/67237353

            QUESTION

            Creating audio player in java
            Asked 2021-Apr-19 at 05:33

            I'm trying to play a simply mp3 file using java. The code I have written is:

            ...

            ANSWER

            Answered 2021-Apr-19 at 05:33

            Your example can play midi, wav files, maybe some others, but not mp3, because of patent issues, but If you can use JavaFX, then you can play mp3 like this:

            Source https://stackoverflow.com/questions/67153254

            QUESTION

            Get max number of Clips that the Mixer can support?
            Asked 2021-Apr-06 at 17:27

            I'm opening multiple audio files as instances of Clip by calling the following method for each audio file:

            ...

            ANSWER

            Answered 2021-Apr-06 at 17:27

            I'm going to venture a guess that the issue about a line being unavailable is due to an attempt reuse a line. AFAIK, you can open as many Clips as you want (assuming you don't run out of memory).

            How many can play at the same time is another question. I think it depends a great deal on the individual PC. I think the last time I tried this was several years ago on an older laptop and it could handle something like 30 Clips or half that many SourceDataLines playing at once.

            If you are trying to play a sound file as a Clip concurrently (making many copies of the same audio resource), you might be interested in looking at AudioCue. This makes use of a plan where the source data is stored just once in memory, but multiple cursors (each a playing instance) are managed, and the outputs from the cursors are merged into a single SourceDataLine. Some very complex soundscapes can be built with this tool.

            Source https://stackoverflow.com/questions/66957141

            QUESTION

            Javascript | MediaRecorder API - Recorded video file on Desktop does not play in Mobile
            Asked 2021-Apr-01 at 11:39

            I am trying to develop a canva-like Insta story creator using Canvas and MediaRecorder The app is working perfectly on a desktop browser - I am able to download the file, and play it on desktop. However, when I send that file to my mobile, it doesn't play(even on Insta). I figure this is an issue with codecs - but don't know how to solve the same.

            This is the function that handles the mediaRecorderAPI

            Is there any mime type that I can use, that is universal and can play for any device?

            ...

            ANSWER

            Answered 2021-Apr-01 at 11:39

            Figured this out: Different browsers use different transcoding. Insta only accepts MP4 transcoding. Hence, you need to use either a transcoder on the frontend(ffmpeg.js or wasm version of ffmpeg) or send your data to backend and handle there(which I ended up doing)

            Source https://stackoverflow.com/questions/65991139

            QUESTION

            Difference in audio sample size between AVCodecContext->frame_size and AVFrame->nb_samples
            Asked 2021-Mar-20 at 11:30

            I am playing with ffmpeg to understand audio data, but I see there is a difference between audio data, AVCodecContext->frame_size shows it to be 1152, but the value I get fromAVFrame->nb_samples shows it to be 47. Both data fields describes the same thing i.e no of samples in an audio frame per channel, then why is there a difference. For reference I m pasting the AVFrame and AVCodecContext Object, which are huge, but also it will give you any information you wanted

            AVCodecContext

            ...

            ANSWER

            Answered 2021-Mar-20 at 11:30

            For MP3 the first 1105 samples are decoder delay, leaving 47 samples returned out of 1152. See How to compute the number of extra samples added by LAME or FFMPEG

            Source https://stackoverflow.com/questions/66720676

            QUESTION

            libvlc - http stream error: local stream 1 error: Cancellation (0x8)
            Asked 2021-Mar-18 at 03:40

            I encounter a bug with libvlc where attempting to play video from YouTube result in error http stream error: local stream 1 error: Cancellation (0x8)

            Context

            C++ code:

            ...

            ANSWER

            Answered 2021-Mar-18 at 03:40

            Since Youtube URLs are not like other URLs for VLC, in the sense that you give a HTML URL to VLC, and that VLC needs to parse that HTML page and reconstruct the actual video URL (that's what the lua script does), you cannot give the HTML URL directly to the mediaplayer as a media.

            Create a media from the HTML URL, as you have done.

            Source https://stackoverflow.com/questions/66657387

            QUESTION

            AWS Polly v3: SynthesizeSpeech 'mp3', AudioStream not instanceOf Buffer
            Asked 2021-Mar-11 at 15:08

            I am converting AWS Polly code in Javascript (node) from v2 to v3.

            When using AWS Polly v3 synthesizeSpeech(), how can I convert the returned data.AudioStream to an instance of Buffer?

            The following JavaScript code works using v2, and returned audio.AudioStream is an instance of Buffer:

            ...

            ANSWER

            Answered 2021-Mar-04 at 11:26

            Now is a ReadableStream object.

            Source https://stackoverflow.com/questions/65959928

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

            Vulnerabilities

            No vulnerabilities reported

            Install audiostream

            You can install using 'npm i audiostream' or download it from GitHub, npm.

            Support

            Bug reports are always welcome! Please file them as issues on the github page: https://github.com/nickdesaulniers/audiostream/issues. Do NOT just fork and send a pull request!!! If there is an issue in the issue tracker that you want to tackle, and it has been accepted, feel free to to fork, and issue a pull request. Otherwise, please first create an issue, and wait for it to be approved. This will allow you to 'fail sooner,' as in have your patch declined before you write it, if it’s not in line with the goals of this project.
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            Install
          • npm

            npm i audiostream

          • CLONE
          • HTTPS

            https://github.com/nickdesaulniers/audiostream.git

          • CLI

            gh repo clone nickdesaulniers/audiostream

          • sshUrl

            git@github.com:nickdesaulniers/audiostream.git

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