I'm using get user media & media recorder api to record users video.
I would like to create a button to clear the last part of the recorded stream since he paused the recording.
I didn't find anyway to do that!
If someones has an idea it would be really helpful !!
PS: if you want an example try TikTok delete option when you pose the recording
MediaRecorder delivers its coded output to the ondataavailable event handler you provide. Unfortunately that data is delivered to your event handler as Blob objects.
You can extract the Blob data into an ArrayBuffer with a variety of methods. But then you'll have to cope with the ebml / matroska / webm boxing format, not to mention the difference between key frames and delta frames in the video stream, and similar sample-buffer boundary issues in the audio stream(s).
What you want to do is going to be difficult-to-impossible in a pure HTML5 / Javascript implementation.
Related
I am working on a project in JavaScript, and I need to do a fairly strange task. I am not sure how to achieve it. I have looked into the most popular libraries for audio, but it doesn't seem to be an easy way to just export a created file fast, without recording it in real-time. I might be wrong.
I am going to have some data as JSON or in CSV format with numbers in each row. That number corresponds to seconds elapsed. This data tells me when a certain audio file needs to be played. The audio file is just a 5-second long clip of beeps.
I need to generate a long audio file that is just silent, but where the audio clip plays when "instructed" by the data. So it could play at 10 seconds, 45 seconds, 267 seconds, etc.
The finished audio file could be over an hour long. I was hoping to create a system where I could just select an audio file and a data file from my computer, click a button, let it process, and then download the finished file.
I hope what I want to do is not unclear. Can I use the Web Audio API for this, or do I need a library? I am stuck at the first part of the process, namely what to use and how to "create" a file from nothing.
I think you could try MediaRecorder from MediaStream Recording API - it will let you create a recording function (something like this):
const stream = navigator.mediaDevices.getUserMedia({ audio: true });
const mediaRecorder = new MediaRecorder(stream);
// then you can get your number from JSON / CSV file and do next
mediaRecorder.start();
await sleep(YOUR_NUMBER_FROM_FILE);
mediaRecorder.stop();
The example above just creates audio that will be last concrete seconds.
You can load your existing files of audio and try it play in the background. I suppose MediaRecorder will record them too.
See MDN Web Docs about MediaRecorder
To summarize - Web Audio API and MediaStream Recording API will be enough for your case.
I hope this will help you in some ways.
I'm using howler.js to play audio on my website, but because I am downloading audio through direct links there are differing load times depending on the length of the audio. I was wondering if there was a way to see how much of the file has been downloaded and display it to the user so that they know an approximate time it would take for it to finish loading. An example of how I am downloading audio is shown below.
lvlHowlWEB[eps] = new Howl({
src:["https://dl.dropboxusercontent.com/s/hb7gw9zqvdptdzq/Potat2k18.mp3?dl=0"],
html5: false,
});
(Note: Audio will not begin playback until after it has loaded completely. I am just looking to see if there's a way to view how much longer it will take to be completely downloaded)
Any help would be appreciated!
You could consider using a ServiceWorker to intercept the HTTP requests and show progress updates as they download. https://fetch-progress.anthum.com/
Alternatively, you could use fetch() and show download progress, then convert the downloaded data into a Blob for URL.createObjectURL(), which could then be used for Howler's src property.
I'm working with the first example on Visualizations with Web Audio API.
My goal is to use a slider to see the waveform at different times without playing the audio.
However, it seems that the audio must be playing in order for analyser.getByteTimeDomainData(dataArray) to get the data at the current time. I set the audioElement.currentTime, but while paused, analyser.getByteTimeDomainData(dataArray) just returns [128, 128, ..., 128].
I'd just like to get the data for a specific time while the audio is paused. Is that possible?
getByteTimeDomainData is used for real-time audio analysis, in order to generate waveform without playing you need to decode the entire audio file using audio context method decodeAudioData and get audio data (Float32Array) from the decoded audio buffer using the getChannelData method which you would process to get probably average sample value for each canvas pixel and render it.
Yes, the audio context must be running for the data to change in an AnalyserNode. I can't think of any other way with WebAudio.
Here is my problem : I want to play a large video file (3.6Gb) stored in a S3 bucket, but it seems the file is too big and the page crash after 30sec of loading.
This is my code to play the video :
var video = document.getElementById("video");
const mediaSource = new MediaSource();
video.src = URL.createObjectURL(mediaSource);
mediaSource.addEventListener('sourceopen', sourceOpen, { once: true });
function sourceOpen() {
URL.revokeObjectURL(video.src);
const sourceBuffer = mediaSource.addSourceBuffer('video/mp4; codecs="avc1.f40028"');
fetch('URL_TO_VIDEO_IN_S3')
.then(response => response.arrayBuffer())
.then(data => {
// Append the data into the new sourceBuffer.
sourceBuffer.appendBuffer(data);
})
.catch(error => {
});
}
I saw that blob URL could be a solution but it didn't work well with my URL.
Take my answer with a grain of salt as I am no expert. However, I am working on something very similar at the moment.
I suspect your issue is that you're attempting to load the entire resource (video file) into the browser at once. An object URL for a file size that exceeds a gigabyte is extremely large.
What you need to do is use the readable stream from the body of your fetch request to process the video file chunk-by-chunk. So as long as you aren't confined to working in the safari browser, you should be to use both the Readable and Writeable Stream classes natively in the browser.
These two classes allow you to form what's called a pipe. In this case, you are "piping" data from the readable stream in your fetch request to a writable stream that you create which is then used as the underlying source of data for your media source extension and it's respective source buffers.
A stream pipe is very special in that it exhibits what's called backpressure. You should definitely look this term up, and read about what it means. In this case, it means the browser will not request more data once it has enough to meet its needs for video playback, the exact amount it can hold at once is specified by you the programmer through something called a "highwater mark" (you should also read about this).
This allows you to control when and how much data the browser is requesting from your (on going) fetch request.
NOTE: When you use .then(response => response.arrayBuffer()) you are telling the browser to wait for the entire resource to come back and then turn the response into an array buffer.
OPTION 1
Use CloudFront to create RTMP distribution to these resources.
It will distribute your video in streaming way.
Create an RTMP distribution to speed up distribution of your streaming media files using Adobe Flash Media Server's RTMP protocol.
Please note that HTML5 does not support RTMP format by default (without flash).
Check here for options
JWPlayer supports RTMP playback using flash. SO Question
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OPTION 2
Use Elastic Transcoder to create HLS video (.m3u8 format). Again same JWPlayer can handle it in ease.
Also it's mostly supported in native HTML5. Check compatibility with H.264
I wonder if it is possible to create a html5 video on the fly. Some of you may noticed the new webrtc and its behavior with the video tag.
navigator.webkitGetUserMedia('video', gotStream, noStream);
function gotStream(stream) {
video.src = webkitURL.createObjectURL(stream);
}
what exactly is that "stream" in gotStream(stream) what is that "interface" looks like so i can generate one of my own? May it be by computing things or by just receiving data from server to display the video. Secound how do i get the data out of this "stream"? So i can read it from one users webcam to send it to my server and let it pass through the receiving user. Binary data transmission is no topic of my question, i already have this working.
I just need the data from the "stream" from one user and reconstruct that "stream" on the target user who wanna see user ones webcam.
Any further information on "where to get these infos by my self" (API Docu sort of) would be also very helpful, cuz i cant find any.
I am aware of the PeerConnection stuff, so no need to mention it here. Cuz beside that webcam stuff i would love to pipe dynamically generated videos from my server to the client or make some sort of video transmitting over dynamic changeable bandwidth with ffmpeg etc. but for this i need to pipe that data to that video element
You may want to look at Whammy: http://antimatter15.com/wp/2012/08/whammy-a-real-time-javascript-webm-encoder/.
For now, you can periodically copy a video element screen to a canvas, then save that canvas to build a video. Whammy ties together webp images generated from a canvas into a webm file, taking advantage of the similarities of the webp (image) and webm (video) format.
You could generate other images and stitch them together in the same manner. Note that this does not support audio.