I'm making a game that changes some of it's object depending on what music is playing. After each song has ended I want my audio context to load in a new source and analyze that. However whenever I tried to do that I've gotten the error that an audio object or buffer can't be called twice.
After some researching I learned that ctx.createMediaElementSource(MyHTML5AudioEl) lets you create a sourceNode that takes the source from a HTML5 object. With this I was able to loop through different song.
However for my game I need to play/analyze a 30 seconds "remote url" that comes out of the Spotify API. I might be wrong but ctx.createMediaElementSource(MyHTML5AudioEl)does not allow you to analyze a source that is on a remote site.
Also the game needs to work on Mobile Chrome, which createMediaElementSource(MyHTML5AudioEl) does not seem to work on.
I might be on the completely wrong path here but my main question is:
How can I switch remote songs urls in web audio api. With it being compatible with mobile chrome.
Thanks!
First, as you found out, you can't set the buffer again for an AudioBufferSource. The solution is to create a new one. AudioBufferSources are intended to be light-weight objects that you can easily create and use.
Second, in Chrome 42 and newer, createMediaElementSource requires appropriate CORS access so you have to make sure the remote url sends the appropriate headers and you set the crossOrigin attribute appropriately.
Finally, Mobile Chrome currently does not pass the data from an audio element through createMediaElementSource.
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I'm really stuck trying to understand the best way to stream real time output of ffmpeg to a HTML5 client using node.js, as there are a number of variables at play and I don't have a lot of experience in this space, having spent many hours trying different combinations.
My use case is:
1) IP video camera RTSP H.264 stream is picked up by FFMPEG and remuxed into a mp4 container using the following FFMPEG settings in node, output to STDOUT. This is only run on the initial client connection, so that partial content requests don't try to spawn FFMPEG again.
liveFFMPEG = child_process.spawn("ffmpeg", [
"-i", "rtsp://admin:12345#192.168.1.234:554" , "-vcodec", "copy", "-f",
"mp4", "-reset_timestamps", "1", "-movflags", "frag_keyframe+empty_moov",
"-" // output to stdout
], {detached: false});
2) I use the node http server to capture the STDOUT and stream that back to the client upon a client request. When the client first connects I spawn the above FFMPEG command line then pipe the STDOUT stream to the HTTP response.
liveFFMPEG.stdout.pipe(resp);
I have also used the stream event to write the FFMPEG data to the HTTP response but makes no difference
xliveFFMPEG.stdout.on("data",function(data) {
resp.write(data);
}
I use the following HTTP header (which is also used and working when streaming pre-recorded files)
var total = 999999999 // fake a large file
var partialstart = 0
var partialend = total - 1
if (range !== undefined) {
var parts = range.replace(/bytes=/, "").split("-");
var partialstart = parts[0];
var partialend = parts[1];
}
var start = parseInt(partialstart, 10);
var end = partialend ? parseInt(partialend, 10) : total; // fake a large file if no range reques
var chunksize = (end-start)+1;
resp.writeHead(206, {
'Transfer-Encoding': 'chunked'
, 'Content-Type': 'video/mp4'
, 'Content-Length': chunksize // large size to fake a file
, 'Accept-Ranges': 'bytes ' + start + "-" + end + "/" + total
});
3) The client has to use HTML5 video tags.
I have no problems with streaming playback (using fs.createReadStream with 206 HTTP partial content) to the HTML5 client a video file previously recorded with the above FFMPEG command line (but saved to a file instead of STDOUT), so I know the FFMPEG stream is correct, and I can even correctly see the video live streaming in VLC when connecting to the HTTP node server.
However trying to stream live from FFMPEG via node HTTP seems to be a lot harder as the client will display one frame then stop. I suspect the problem is that I am not setting up the HTTP connection to be compatible with the HTML5 video client. I have tried a variety of things like using HTTP 206 (partial content) and 200 responses, putting the data into a buffer then streaming with no luck, so I need to go back to first principles to ensure I'm setting this up the right way.
Here is my understanding of how this should work, please correct me if I'm wrong:
1) FFMPEG should be setup to fragment the output and use an empty moov (FFMPEG frag_keyframe and empty_moov mov flags). This means the client does not use the moov atom which is typically at the end of the file which isn't relevant when streaming (no end of file), but means no seeking possible which is fine for my use case.
2) Even though I use MP4 fragments and empty MOOV, I still have to use HTTP partial content, as the HTML5 player will wait until the entire stream is downloaded before playing, which with a live stream never ends so is unworkable.
3) I don't understand why piping the STDOUT stream to the HTTP response doesn't work when streaming live yet if I save to a file I can stream this file easily to HTML5 clients using similar code. Maybe it's a timing issue as it takes a second for the FFMPEG spawn to start, connect to the IP camera and send chunks to node, and the node data events are irregular as well. However the bytestream should be exactly the same as saving to a file, and HTTP should be able to cater for delays.
4) When checking the network log from the HTTP client when streaming a MP4 file created by FFMPEG from the camera, I see there are 3 client requests: A general GET request for the video, which the HTTP server returns about 40Kb, then a partial content request with a byte range for the last 10K of the file, then a final request for the bits in the middle not loaded. Maybe the HTML5 client once it receives the first response is asking for the last part of the file to load the MP4 MOOV atom? If this is the case it won't work for streaming as there is no MOOV file and no end of the file.
5) When checking the network log when trying to stream live, I get an aborted initial request with only about 200 bytes received, then a re-request again aborted with 200 bytes and a third request which is only 2K long. I don't understand why the HTML5 client would abort the request as the bytestream is exactly the same as I can successfully use when streaming from a recorded file. It also seems node isn't sending the rest of the FFMPEG stream to the client, yet I can see the FFMPEG data in the .on event routine so it is getting to the FFMPEG node HTTP server.
6) Although I think piping the STDOUT stream to the HTTP response buffer should work, do I have to build an intermediate buffer and stream that will allow the HTTP partial content client requests to properly work like it does when it (successfully) reads a file? I think this is the main reason for my problems however I'm not exactly sure in Node how to best set that up. And I don't know how to handle a client request for the data at the end of the file as there is no end of file.
7) Am I on the wrong track with trying to handle 206 partial content requests, and should this work with normal 200 HTTP responses? HTTP 200 responses works fine for VLC so I suspect the HTML5 video client will only work with partial content requests?
As I'm still learning this stuff its difficult to work through the various layers of this problem (FFMPEG, node, streaming, HTTP, HTML5 video) so any pointers will be greatly appreciated. I have spent hours researching on this site and the net, and I have not come across anyone who has been able to do real time streaming in node but I can't be the first, and I think this should be able to work (somehow!).
EDIT 3: As of IOS 10, HLS will support fragmented mp4 files. The answer
now, is to create fragmented mp4 assets, with a DASH and HLS manifest. > Pretend flash, iOS9 and below and IE 10 and below don't exist.
Everything below this line is out of date. Keeping it here for posterity.
EDIT 2: As people in the comments are pointing out, things change.
Almost all browsers will support AVC/AAC codecs.
iOS still requires HLS. But via adaptors like hls.js you can play
HLS in MSE. The new answer is HLS+hls.js if you need iOS. or just
Fragmented MP4 (i.e. DASH) if you don't
There are many reasons why video and, specifically, live video is very difficult. (Please note that the original question specified that HTML5 video is a requirement, but the asker stated Flash is possible in the comments. So immediately, this question is misleading)
First I will restate: THERE IS NO OFFICIAL SUPPORT FOR LIVE STREAMING OVER HTML5. There are hacks, but your mileage may vary.
EDIT: since I wrote this answer Media Source Extensions have matured,
and are now very close to becoming a viable option. They are supported
on most major browsers. IOS continues to be a hold out.
Next, you need to understand that Video on demand (VOD) and live video are very different. Yes, they are both video, but the problems are different, hence the formats are different. For example, if the clock in your computer runs 1% faster than it should, you will not notice on a VOD. With live video, you will be trying to play video before it happens. If you want to join a a live video stream in progress, you need the data necessary to initialize the decoder, so it must be repeated in the stream, or sent out of band. With VOD, you can read the beginning of the file them seek to whatever point you wish.
Now let's dig in a bit.
Platforms:
iOS
PC
Mac
Android
Codecs:
vp8/9
h.264
thora (vp3)
Common Delivery methods for live video in browsers:
DASH (HTTP)
HLS (HTTP)
flash (RTMP)
flash (HDS)
Common Delivery methods for VOD in browsers:
DASH (HTTP Streaming)
HLS (HTTP Streaming)
flash (RTMP)
flash (HTTP Streaming)
MP4 (HTTP pseudo streaming)
I'm not going to talk about MKV and OOG because I do not know them very well.
html5 video tag:
MP4
webm
ogg
Lets look at which browsers support what formats
Safari:
HLS (iOS and mac only)
h.264
MP4
Firefox
DASH (via MSE but no h.264)
h.264 via Flash only!
VP9
MP4
OGG
Webm
IE
Flash
DASH (via MSE IE 11+ only)
h.264
MP4
Chrome
Flash
DASH (via MSE)
h.264
VP9
MP4
webm
ogg
MP4 cannot be used for live video (NOTE: DASH is a superset of MP4, so don't get confused with that). MP4 is broken into two pieces: moov and mdat. mdat contains the raw audio video data. But it is not indexed, so without the moov, it is useless. The moov contains an index of all data in the mdat. But due to its format, it can not be 'flattened' until the timestamps and size of EVERY frame is known. It may be possible to construct an moov that 'fibs' the frame sizes, but is is very wasteful bandwidth wise.
So if you want to deliver everywhere, we need to find the least common denominator. You will see there is no LCD here without resorting to flash
example:
iOS only supports h.264 video. and it only supports HLS for live.
Firefox does not support h.264 at all, unless you use flash
Flash does not work in iOS
The closest thing to an LCD is using HLS to get your iOS users, and flash for everyone else.
My personal favorite is to encode HLS, then use flash to play HLS for everyone else. You can play HLS in flash via JW player 6, (or write your own HLS to FLV in AS3 like I did)
Soon, the most common way to do this will be HLS on iOS/Mac and DASH via MSE everywhere else (This is what Netflix will be doing soon). But we are still waiting for everyone to upgrade their browsers. You will also likely need a separate DASH/VP9 for Firefox (I know about open264; it sucks. It can't do video in main or high profile. So it is currently useless).
Thanks everyone especially szatmary as this is a complex question and has many layers to it, all which have to be working before you can stream live video. To clarify my original question and HTML5 video use vs flash - my use case has a strong preference for HTML5 because it is generic, easy to implement on the client and the future. Flash is a distant second best so lets stick with HTML5 for this question.
I learnt a lot through this exercise and agree live streaming is much harder than VOD (which works well with HTML5 video). But I did get this to work satisfactorily for my use case and the solution worked out to be very simple, after chasing down more complex options like MSE, flash, elaborate buffering schemes in Node. The problem was that FFMPEG was corrupting the fragmented MP4 and I had to tune the FFMPEG parameters, and the standard node stream pipe redirection over http that I used originally was all that was needed.
In MP4 there is a 'fragmentation' option that breaks the mp4 into much smaller fragments which has its own index and makes the mp4 live streaming option viable. But not possible to seek back into the stream (OK for my use case), and later versions of FFMPEG support fragmentation.
Note timing can be a problem, and with my solution I have a lag of between 2 and 6 seconds caused by a combination of the remuxing (effectively FFMPEG has to receive the live stream, remux it then send it to node for serving over HTTP). Not much can be done about this, however in Chrome the video does try to catch up as much as it can which makes the video a bit jumpy but more current than IE11 (my preferred client).
Rather than explaining how the code works in this post, check out the GIST with comments (the client code isn't included, it is a standard HTML5 video tag with the node http server address). GIST is here: https://gist.github.com/deandob/9240090
I have not been able to find similar examples of this use case, so I hope the above explanation and code helps others, especially as I have learnt so much from this site and still consider myself a beginner!
Although this is the answer to my specific question, I have selected szatmary's answer as the accepted one as it is the most comprehensive.
Take a look at JSMPEG project. There is a great idea implemented there — to decode MPEG in the browser using JavaScript. Bytes from encoder (FFMPEG, for example) can be transfered to browser using WebSockets or Flash, for example. If community will catch up, I think, it will be the best HTML5 live video streaming solution for now.
One way to live-stream a RTSP-based webcam to a HTML5 client (involves re-encoding, so expect quality loss and needs some CPU-power):
Set up an icecast server (could be on the same machine you web server is on or on the machine that receives the RTSP-stream from the cam)
On the machine receiving the stream from the camera, don't use FFMPEG but gstreamer. It is able to receive and decode the RTSP-stream, re-encode it and stream it to the icecast server. Example pipeline (only video, no audio):
gst-launch-1.0 rtspsrc location=rtsp://192.168.1.234:554 user-id=admin user-pw=123456 ! rtph264depay ! avdec_h264 ! vp8enc threads=2 deadline=10000 ! webmmux streamable=true ! shout2send password=pass ip=<IP_OF_ICECAST_SERVER> port=12000 mount=cam.webm
=> You can then use the <video> tag with the URL of the icecast-stream (http://127.0.0.1:12000/cam.webm) and it will work in every browser and device that supports webm
I wrote an HTML5 video player around broadway h264 codec (emscripten) that can play live (no delay) h264 video on all browsers (desktop, iOS, ...).
Video stream is sent through websocket to the client, decoded frame per frame and displayed in a canva (using webgl for acceleration)
Check out https://github.com/131/h264-live-player on github.
This is a very common misconception. There is no live HTML5 video support (except for HLS on iOS and Mac Safari). You may be able to 'hack' it using a webm container, but I would not expect that to be universally supported. What you are looking for is included in the Media Source Extensions, where you can feed the fragments to the browser one at a time. but you will need to write some client side javascript.
Take a look at this solution.
As I know, Flashphoner allows to play Live audio+video stream in the pure HTML5 page.
They use MPEG1 and G.711 codecs for playback.
The hack is rendering decoded video to HTML5 canvas element and playing decoded audio via HTML5 audio context.
How about use jpeg solution, just let server distribute jpeg one by one to browser, then use canvas element to draw these jpegs?
http://thejackalofjavascript.com/rpi-live-streaming/
I am trying to make a webapp that will load a page from a remote server, but allow the user to play audio from files that are on their local drive (not downloaded from the remote server). I am able to get this to work, but I also need it to save what the user has done for subsequent visits. For example: the user loads a page, clicks a "choose file" button, selects an mp3, and plays it. The user then closes the browser, opens it again, returns to the page, and is able to play the same audio without having to select it again.
I understand that the audio playback is separate from the saving of the user's selection, but in this case one seems to dictate the other.
I am able to get the select-and-play functionality to work with this:
<html><body>
<script type='text/javascript'>
function handleFiles(files){
var file = window.URL.createObjectURL(files[0]);
document.getElementById('audioPlayer').src = file;
}
</script>
<audio id='audioPlayer' controls ></audio>
<input type='file' id='selectedFile'
onchange='handleFiles(this.files)' />
</body></html>
...but I do not know how to store the selected file data in a way that I can automatically load it on the next visit. What can I use to store that file location (or even the whole file itself if it comes to it) so that I can still play the audio without the user selecting the file again?
I kind of suspect that saving the local file url somehow may not be possible for security reasons, since auto-playing a file from the local file system without user interaction could be bad news.
File handles from File open dialog are not recycleable across different page load sessions.
The best you can do this that you copy audio data to a HTML5 localStorage and play it from there. Or upload the data to your server and play it from there.
http://docs.webplatform.org/wiki/apis/web-storage/Storage/localStorage
localStorage is limited to few megabytes depending on the browser.
At this time, Mikko's answer is the correct answer for my question, but I thought I'd share a possible alternative for anyone else who comes across this thread:
The FileSystem API looks like it would perfectly suit my needs in this case, but at the time of this writing, it is only supported in Chrome. If audio playback is a minor add-on feature to your webapp though, this might be an option to give Chrome users a better experience and other users would just be unaware that they're missing out.
In this HTML5 Rocks article, the author shows how to use it, including how to copy user-selected files into a local disk sandbox and how to get a url (needed in my case to audio playback) to those files.
Hi my name is Tamer and this is very first post in stackoverflow as far as I know. :)
I've been searching this for an issue of mine for a while but haven't found any solutions yet. I want to forward my radio broadcast from it's original host to my embedded flash player once the play button is hit. Is it possible via action script (flash - as3) ? My broadcast is in Icecast format, and I'm playing mp3 but my mount point is not exactly /listen.mp3. It has some more extention like ?auth and some random fixed stuff (I mean, there are some other things at the end of the auth part, but it's same all the time like a fixed link).
Thanks for your help and time!
Yes, you can play your stream with Flash. It doesn't matter what your mount point is. Just plug in the real URL for the stream. The file name extension and what not doesn't matter.
When using the above answer high memnory use may be an issue for your viewers as the "Sound" variable downloads the data while playing.
Would be better off using progressive play/download but would have to change your streaming format from mp3 as it only allows for M4A streaming (AAC audio data in an MP4 container).
var netConnection:NetConnection = new NetConnection();
netConnection.connect(null);
var netStream:NetStream = new NetStream(netConnection);
netStream.play(“http://your.stream.url/stream.m4a”);
Just my 2cents :)
I need to record a video through user browser using input from camera and microphone and send to my server. Since html5 still doesn't make that magic happen, I'm looking for flash solutions.
Do I really need some flash media server to do that, or can I do a POST request?
I want to get both inputs(webcam and microphone), put them in a .flv and send to my server.
I've seen some implementations using bytearrays to record and send, audio and video separated. The problem is that it generates a series of synchronization problems when you try to compose them in a single file.
If you're still looking for a solution check out:
http://framebase.io
They have an embed-able recording widget that can transcode the videos automatically. I'd check out the docs, but on success, you can run an API call to check the status of transcoding and download it to your server or you can just use your own S3 bucket.