audiostream | Stream and transcode your music library over HTTP | Audio Utils library
kandi X-RAY | audiostream Summary
kandi X-RAY | audiostream Summary
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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Currently covering the most popular Java, JavaScript and Python libraries. See a Sample of audiostream
audiostream Key Features
audiostream Examples and Code Snippets
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
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
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);
// 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
Trending Discussions on audiostream
QUESTION
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 02:42This 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
QUESTION
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 02:11I fixed it by increasing a function where I capture the audio from the microphone
QUESTION
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 10:52Does 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
andcos
. 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;
QUESTION
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 22:29In 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.
QUESTION
I'm trying to play a simply mp3 file using java. The code I have written is:
...ANSWER
Answered 2021-Apr-19 at 05:33Your 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:
QUESTION
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 17:27I'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.
QUESTION
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 11:39Figured 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)
QUESTION
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 11:30For 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
QUESTION
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)
C++ code:
...ANSWER
Answered 2021-Mar-18 at 03:40Since 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.
QUESTION
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 11:26Now is a ReadableStream object.
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