I have audio files in OGG format and embed them with:
<div class="audio-container">
<audio controls>
<source src="/u/audiofile.ogg">
</audio>
</div>
I play the embedded file with Chrome (same issue with Internet Explorer/Edge).
If the file is about 3 seconds long, the progressbar/playbar appears at the beginning as expected:
If the file is longer, in the example 25 seconds, the progressbar/playbar does not work, that means: Clicking on the playbar does not jump to the audio time:
However, after 22 seconds (3 seconds before the end), the playbar suddenly appears and the user can jump through the audio times by clicking on the playbar.
As we can see here, only after the entire OGG file has played (not only loaded, but played) the 206 partial content request suddenly works.
Of course I want that the playbar shows the entire audio length and that navigate-clicking is possible from the beginning.
How to achieve this?
Update:
I have tried a Javascript plugin (green audio player). The player shows up like this:
There is Infinity showing up. So I guess the file length is not sent by the server. And this is strange, because the file is linked directly (file system location).
The response headers look like this (for the 25 seconds file):
My Apache server replies with status code 200 for the first load. When hitting the playbar for a video it does a 206 Partial content request as needed. However, this does not happen when clicking on the audio playbar in Chrome. No request is sent.
Contrary it sends the Accept-Ranges: bytes which should enable seeking through the file:
The Accept-Ranges header indicates that the server accept range requests for a resource. The unit used by the remote web servers is in bytes. This header tells us that either the server supports download resume or downloading files.
I assume that one way of solving this issue is to always supply OGG files with 206 Partial content, already with the first request. But I could not find out yet how to enable this in the Apache server.
Notes:
I read that Apache does 206 by default, but not if caching/GZIP is enabled. So I disabled caching in htaccess (SetEnvIfNoCase Request_URI . (?:mp4|mp3|ogg|webm)$ no-gzip dont-vary) but it still does not work.
Update:
I tried with another OGG file (test.ogg). This one plays and jumping with the playbar is possible without problems!
The OGG files I am using are created by recording them browser-side (Javascript new MediaRecorder(stream);), then send to the server and stored by PHP as OGG file.
Seeing the file properties of the example file:
Seeing the file properites of the test.ogg file:
Obviously, there is meta data missing with the problematic file!
Follow up: How to save MediaRecorder stream to OGG file in PHP? Including Metadata
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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/
Long version:
I use html5 audio tag to play mp3 files on my website. With Flash I can stream mp3's and secure it for 95%.
With html5 it is easy to find out the mp3 location and just download it from there. Even if I secure it with unique hashes it is not hard to inspect the network tab in chrome and see the mp3 url with hashes.
I was wondering if there are other ways to secure the mp3 from being ripped and if it is worth the time. For example bandcamp does generate unique hashes but it is still very easy to download the mp3. For youtube you got download websites that can proces the flv stream and rip the audio and save it for the user as mp3 format.
The first layer of security I can think of is change the extension of mp3 files to .txt or another common format.
95% of the users don't spot the extension because it is hidden by default on windows and apple. This will prevent the first 95% of the users to spot and play the mp3 file.
Short Version
Any suggestions to prevent users from stealing mp3 files while using html5 audio tag.
Short Answer
No.
Renaming the audio file to .txt is not going to do anything to help the security of your mp3 audio file. If anything, it is going to cause you even more issues, because now, your mp3 audio file is going to be sent with the incorrect MIME type, which may cause issues with the browser's built in audio player.
The best suggestions that I can provide you is:
Make sure that your checking the REFERER http header, make sure that it is coming from the page that has the mp3 player on it.
Protect the mp3 file with a unique hash.
Don't allow the same hash to be downloaded twice*
*Note that even doing this could cause issues, for example, what happens if the user reopens a tab from cache, plays the file again, and the mp3 file is not cached?
And finally, at the end even after your mp3 file is the most protected mp3 file in the history of IIS and Apache -- what is stopping me from just opening up Adobe Audition, and recordinging the audio stream?
Although you are correct about Bandcamp's MP3 audio stream, the mp3 is not as high quality then just a normal download after purchasing an album.
The fact that even Google does not really have any decent protections on it's video streams should say something. A company that generates billions of dollars from video views on YouTube can't even make (or better put -- has not bothered to put in place) any viable methods for protecting their videos.
Kind of.
Grooveshark send a POST request to a server-side script for the MP3 that is being streamed which makes it very difficult to just access and spoof without dynamically creating a POST request yourself - especially seeing as you would have to then attempt to store the audio file that is collected. But you can use the new AudioContext to help solve this for most modern platforms...
I used a great example from HTML5Rocks.com to alter the headers used as follows:
var dogBarkingBuffer = null;
// Fix up prefixing
window.AudioContext = window.AudioContext || window.webkitAudioContext;
var context = new AudioContext();
function loadDogSound(url) {
var request = new XMLHttpRequest();
request.open('POST', url, true);
request.setRequestHeader("Content-type","application/x-www-form-urlencoded");
request.responseType = 'arraybuffer';
// Decode asynchronously
request.onload = function() {
context.decodeAudioData(request.response, function(buffer) {
dogBarkingBuffer = buffer;
}, onError);
}
//this is the encryption key
request.send("key=98753897358975387943");
}
Related
As you can see I am also sending a key value which could possibly be part of a public/private pair too. This should put anyone off attempting to intervene - other than simply recording the MP3 as it's playing, of course, but what could possibly stop that in any environment inside or out of a computer?
You could make the MP3s themselves unattractive. Some ideas:
Don't include album art, album info, etc. in your files (id3 tags). Even better, fill out all id3 tag fields with something like "This file is from myMusicSite.com".
Split your files into several smaller pieces, and then play them in sequence in the browser. Having to download all the individual pieces will make your files much less attractive. You may have problems with gapless playback, not sure how well that's supported though.
Encode and play them as a video, maybe with your logo or something as the video stream. The resulting files won't be much larger, esp. if you use a static image. This will mean that users can't play your files on mp3 players, phones, etc. easily.
Whisper your domain name or website name in the recordings a few times, as mentioned in the comments.
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.
I am seeing, in firebug, that my .ogv video(770KB) is being downloaded in several partial content requests while if I replace this with another video(1.7 MB) that I got from somewhere else then this is downloaded always in a single request.
What may be causing this ?
I think it's just firefox trying to be smart. I would not worry about it.
By reading https://developer.mozilla.org/en/Configuring_servers_for_Ogg_media and https://bugzilla.mozilla.org/show_bug.cgi?id=502894#c1 I think firefox first start downloading the file, then if it finds the file header doesn't contain the duration of the movie, it downloads the last part of the file, which contains the timestamp of the last frame, after that it will continue downloading where it stopped at the beginning of the file.
My guess is the other file has the duration coded in the header somehow, or, if you are hotlinking it from some other server, that the other server is not supporting http-ranges