This is sort of expanding on my previous question Web Audio API- onended event scope, but I felt it was a separate enough issue to warrant a new thread.
I'm basically trying to do double buffering using the web audio API in order to get audio to play with low latency. The basic idea is we have 2 buffers. Each is written to while the other one plays, and they keep playing back and forth to form continuous audio.
The solution in the previous thread works well enough as long as the buffer size is large enough, but latency takes a bit of a hit, as the smallest buffer I ended up being able to use was about 4000 samples long, which at my chosen sample rate of 44.1k would be about 90ms of latency.
I understand that from the previous answer that the issue is in the use of the onended event, and it has been suggested that a ScriptProcessorNode might be of better use. However, it's my understanding that a ScriptProcessorNode has its own buffer of a certain size that is built-in which you access whenever the node receives audio and which you determine in the constructor:
var scriptNode = context.createScriptProcessor(4096, 2, 2); // buffer size, channels in, channels out
I had been using two alternating source buffers initially. Is there a way to access those from a ScriptProcessorNode, or do I need to change my approach?
No, there's no way to use other buffers in a scriptprocessor. Today, your best approach would be to use a scriptprocessor and write the samples directly into there.
Note that the way AudioBuffers work, you're not guaranteed in your previous approach to not be copying and creating new buffers anyway - you can't simultaneously be accessing a buffer from the audio thread and the main thread.
In the future, using an audio worker will be a bit better - it will avoid some of the thread-hopping - but if you're (e.g.) streaming buffers down from a network source, you won't be able to avoid copying. (It's not that expensively, actually.) If you're generating the audio buffer, you should generate it in the onaudioprocess.
Related
In WebRTC we have MediaStream and MediaStreamTrack interfaces.
MediaStreamTrack represents a audio or video stream of a media source. So a consumer like video or audio tag can simply take an MediaStreamTrack object and and get the stream from it. So what is the need for MediaStream interface?
According to official documentation, MediaStream synchronises one or more tracks. Does that mean it combines multiple streams from tracks and produces a single stream so that we have video data with audio?
For example: Does a video tag read the stream from MediaStream object or reads streams from the individual tracks?
This concept is not explained clearly anywhere.
Thanks in advance.
MediaStream has devolved into a simple container for tracks, representing video and audio together (a quite common occurrence).
It doesn't "combine" anything, it's merely a convenience keeping pieces together that need to be played back in time synchronization with each other. No-one likes lips getting out of sync with spoken words.
It's not even really technically needed, but it's a useful semantic in APIs for:
Getting output from hardware with camera and microphone (usually video and audio), and
Connecting it (the output) to a sinc, like the html video tag (which accepts video and audio).
Reconstituting audio and video at the far end of an RTCPeerConnection that belong together, in the sense that they should generally be played in sync with each other (browsers have more information about expectations on the far end this way, e.g. if packets from one track are lost but not the other).
Whether this is a useful abstraction may depend on the level of detail you're interested in. For instance the RTCPeerConnection API, which is still in Working Draft stage, has over the last year moved away from streams as inputs and outputs to dealing directly with tracks, because the working group believes that details matter more when it comes to transmission (things like tracking bandwidth use etc.)
In any case, going from one to the other will be trivial:
var tracks = stream.getTracks();
console.log(tracks.map(track => track.kind)); // audio,video
video.srcObject = new MediaStream(tracks);
once browsers implement the MediaStream constructor (slated for Firefox 44).
I am writing a simple mpeg-dash streaming player using HTML5 video element.
I am creating MediaSource and attaching a SourceBuffer to it. Then I am appending dash fragments into this sourcebuffer and everything is working fine.
Now, what I want to do is, I want to pre-fetch those segments dynamically depending upon current time of the media element.
While doing this there are lot of doubts and which are not answered by MediaSource document.
Is it possible to know how much data sourceBuffer can support at a time? If I have a very large video and append all the fragments into sourcebuffer, will it accommodate all fragments or cause errors or will slow down my browser?
How to compute number of fragments in sourcebuffer?
How to compute the presentation time or end time of the last segment in SourceBuffer?
How do we remove only specific set of fragments from SourceBuffer and replace them with segments with other resolutions? (I want to do it to support adaptive resolution switching run time.)
Thanks.
The maximum amount of buffered data is an implementation detail and is not exposed to the developer in any way AFAIK. According to the spec, when appending new data the browser will execute the coded frame eviction algorithm which removes any buffered data deemed unnecessary by the browser. Browsers tend to remove any part of the stream that has already been played and don't remove parts of the stream that are in the future relative to current time. This means that if the stream is very large and the dash player downloads it very quickly, faster than the MSE can play it, then there will be a lot of the stream that cannot be remove by the coded frame eviction algorithm and this may cause the append buffer method to throw a QuotaExceededError. Of course a good dash player should monitor the buffered amount and not download excessive amounts of data.
In plain text: You have nothing to worry about, unless your player downloads all of the stream as quickly as possible without taking under consideration the current buffered amount.
The MSE API works with a stream of data (audio or video). It has no knowledge of segments. Theoretically you could get the buffered timerange and map to to a pair of segments using the timing data provided in the MPD. But this is fragile IMHO. Better is to keep track of the downloaded and fed segments.
Look at the buffered property. The easiest way to get the end time in seconds of the last appended segments is simply: videoElement.buffered.end(0)
If by presentation time you mean the Presentation TimeStamp of the last buffered frame then there is no way of doing this apart from parsing the stream itself.
To remove buffered data you can use the remove method.
Quality switching is actually quite easy although the spec doesn't say much about it. To switch the qualities the only thing you have to do is append the init header for the new quality to the SourceBuffer. After that you can append the segments for the new quality as usual.
I personally find the youtube dash mse test player a good place to learn.
The amount of data a sourceBuffer can support depends on the MSE implementation and therefore the browser vendor. Once you reached the maximum value, this will of course result in an error.
You cannot directly get the number of segments in SourceBuffer, but you can get the actual buffered time. In combination with the duration of the segments you are able to compute it.
I recommend to have a look in open source DASH player projects like dashjs or ExoPlayer, which implement all your desired functionalities. Or maybe even use a commercial solution like bitdash.
I'm testing a streaming web application that uses MediaSourceAPI. Everything works fine, however when i stream big files (i.e 240MB or more), the buffer of the video has a strange behavior. To be more clear i attached three images you can check. My script creates a mediaSource object, then it calls addSourceBuffer and then it calls appendBuffer many time as there are chunks to append. I think that i do not configure well the buffer and so the mediaSource API use a default value for the buffer length.
Could you help me please?
Visit https://productforums.google.com/forum/#!category-topic/chrome/report-a-problem-and-get-troubleshooting-help/windows8/Stable/0igRzDJQ7ds
There is a max limit on the size of the SourceBuffers, maybe you're exceeding those? When they exceed the limits, the browsers will start evicting buffer segments according to some defined algorithm.
If you are appending as much data to the source buffers as you can, you might want to introduce a limit. E.g. for us, when playing HD video at 4.5mps, we could have a buffer size of about 3-4 minutes before we saw some odd behaviour (e.g. segments being evicted in front of the videos currentTime)
I am using html5 web audio api in my application. Application is simple, I have
BufferSourceNode -> GainNode -> lowpass filter -> context.destination
Now I want to save the output after applying the filters. So I decided to add recorder before
context.destination. But this doesn't work, it gives some noise sound while playing the audio, though my recorder records filter effects successfully.
Am I doing it in right way or is there any better way to do this?
Two things:
1) if you are going to use the buffer anyway - even if you're not() - you might want to consider using an OfflineAudioContext (https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html#OfflineAudioContext-section). OACs can run faster than real-time, so you don't need to "record" it in real-time; you set up your nodes, call startRendering(), and the oncomplete event hands you an audiobuffer. () If you still want a .WAV file, you can pull the WAV encoding function out of Recordjs and use it to encode an arbitrary buffer.
2) That sounds like an error in your code - it should work either way, without causing extra noise. Do you have a code sample you can send me?
I have a server that generates pngs very rapidly and I need to make this into a poor-man's video feed. Actually creating a video feed is not an option.
What I have working right now is a recursive loop that looks a little like this (in pseudo-code):
function update() {
image.src = imagepath + '?' + timestamp; // ensures the image will update
image.onload = function () {update()};
}
This works, however after a while, it crashes the browser (Google Chrome, after more than 10 minutes or so). These images are being updated very frequently (several times a second). It seems the images are caching, which causes the browser to run out of memory.
Which of these solutions would solve the problem while maintaining fast refresh:
HTML5 canvas with drawImage
HTML5 canvas with CanvasPixelArray (raw pixel manipulation)
I have access to the raw binary as a Uint8Array, and the image isn't too large (less than 50 kb or so, 720 x 480 pixels).
Alternatively, is there anyway to clear old images from the cache or to avoid caching altogether?
EDIT:
Note, this is not a tool for regular users. It's a tool for diagnosing analog hardware problems for engineers. The reason for the browser is platform independence (should work on Linux, Windows, Mac, iPad, etc without any software changes).
The crashing is due to http://code.google.com/p/chromium/issues/detail?id=36142. Try creating object URLs (use XHR2 responseType = "arraybuffer" along with BlobBuilder) and revoking (using URL.revokeObjectURL) the previous frame after the next frame is loaded.
Edit: You really should be processing these into a live low-fps video stream on the server side, which will end up giving you greatly decreased latency and faster load times.
#Eli Grey seems to have identified the source of your crashing. It looks like they have a fix in the works, so if you don't want to modify your approach hopefully that will be resolved soon.
With regard to your other question, you should definitely stick with drawImage() if you can. If I understand your intention of using the CanvasPixelArray, you are considering iterating over each pixel in the canvas and updating it with your new pixel information? If so, this will be nowhere near as efficient as drawImage(). Furthermore, this approach is completely unnecessary for you because you (presumably) do not need to reference the data in the previous frame.
Whether fortunately or not, you cannot directly swap out the internal CanvasPixelArray object stored within an HTML5 canvas. If you have a properly-formatted array of pixel data, the only way you can update a canvas element is by calling either drawImage() or putImageData(). Right now, putImageData() is much slower than drawImage(), as you can see here: http://jsperf.com/canvas-drawimage-vs-putimagedata. If you have any sort of transparency in the frames of your video, you will likely want to clear the canvas and then use drawImage() (otherwise you could see through to previous frames).
Having said all that, I don't know that you really even need to use a canvas for this. Was your motivation for looking into using a canvas so that you could avoid caching (which now doesn't seem to be the underlying issue for you)?
If the "movie" is data-driven (ie. based on numbers and calculations), you may be able to push MUCH more data to the browser as text and then have javascript render it client-side into a movie. The "player" in the client can then request the data in batches as it needs it.
If not, one thing you could do is simply limit the frames-per-second (fps) of the script, possibly a hard-coded value, or a slider / setable value. Assuming this doesn't limit the utility of the tool, at the very least it would let the browser run longer w/o crashing.
Lastly, there are lots of things that can be done with headers (eg. in the .htaccess file) to indicate to browsers to cache or not cache content.
iPad, you say ?.. Nevertheless, i would advice using Flash/video or HTML5/video.
Because WebKit is very easily crashed with even moderate influx of images, either just big images or just a huge number of small ones..
From the other side, XHR with base64 image data or pixel array MIGHT work. I have had short polling app, which was able to run for 10-12 hours with XHR polling server every 10 seconds.
Also, consider delta compression, - like, if its histogram with abscissa being time scale - you can only send a little slice from the rigth, - of course, for things like heat-maps, you cannot do that.
These images are being updated very frequently (several times a
second).
.. if its critical to update at such a high rate - you MUST use long polling.