I am trying a build a realtime monitoring system for high frequency data. To increase the performance, I used the extendData property of dcc.Graph() and websocket. So that, the brouser does not need to send request to get data.
I found that it still not increasing the performance as expected. The reason I found is, from the browser, I see (by inspecting network from browser) after some miliseconds browser is still sendng request and the initiator is the dash_renderer.
This picture is for a vanilla example just to show even for a textbox example the http request goes on and on. And for my real time websocket dashboard the frequency of requests get very high.
My question is:
What dash_renderer do?
why it is sending http request?
And how to stop that?
If you run Dash in Debug mode, it has a feature called Hot Reloading which regularly (every 3 seconds by default) checks for changes to your codebase and updates your running app if it finds any. That check for updated code is what you're seeing in the network inspection.
To turn it off, either don't run in debug mode or explicitly set dev_tools_hot_reload to False like so:
app.run_server(debug=True, dev_tools_hot_reload=False)
Although it is late, After some experience, my realization is dash is not designed to work with websocket. It uses call-backs which actually sends requests to server and in server, the callback function (which is python) send back some result.
These call-backs are designed to send HTTP request to server.
For high speed data, the websocket should be used with extendTrace method of plotly.js in client side.
I have a simple setup for Desktop Capturing using html5 libraries.
This includes a simple webpage and a chrome-extension. I am using
Extension to get the sourceId
Using the sourceId I call navigator.mediaDevices.getUserMedia to get the MediaStream
This MediaStream is then fed into an instance of MediaRecorder for recording.
This setup works most of the times, but a few times I see that requestData() on MediaRecorder instance returns blob with empty data consistently. I am clueless as to what can cause a running setup to start misbehaving sometimes.
Some weird behaviour that I noticed in the bad state:
When I try to close/refresh the window it doesn't respond.
The MediaStreamTrack object in Step 2) above is 'live' but as soon as I go to Step 3) it becomes 'muted'.
There's no pattern to it, sometimes it even happens when I request for the MediaStreams the very first time(which rules out the possibility that there could be some dangling resources eating up the contexts)
Is there anything that I am doing wrong and am unaware of? Any help/pointers would be highly appreciated!
We are using Pusher on mobile devices (using html webview + javascript).
I am connecting to Pusher using the following script (loaded async to the page)
http://js.pusher.com/2.1/pusher.min.js.
and then initialise Pusher object:
var externalTrackingProvider = new Pusher(config.key, { encrypted: true });
var connection = externalTrackingProvider.connection;
connection.bind('connected', function() {
console.log("connected");
});
The process works well but the time between "new Pusher" and "connected" event can take up to 15!!! seconds on mobile devices. The average time between init and connected is 5-10 seconds. (the js file itself is pre-loaded so this is the actual connection time).
Our testing is done using high speed wifi network so this is not the issue here.
Is this a knows Pusher issue? Anything to be done in order to fix this?
EDIT
Following Mike's advise, I tested "http://test.pusher.com" using our webview on Samsung Galaxy S4, high speed wifi connection. The results are avarage of 3-4 seconds between "connecting" and "connected". Attached screenshot.
Edit2:
After sending the logs to Pusher it appears that Android webview does not support Websockets, which causes a js fallback which results in poor performance.
The solution should be some kind of webview websocket library
I'll try and update if it is working
It should all be done in well under a second. You can use test.pusher.com with the same browser and network connection to see the messages passed back and forth and at what stage the latency is being introduced. Some fallbacks where websocket connections fail require the loading of further code, may involve a timeout to consider the initial attempt failed and require more round-trips than a websocket in order to create a connection.
Is it possible for a Chrome extension to listen for streaming audio from any of the browser's tabs? I would like to capture the streaming audio data and then analyse it.
Thanks
You could try 3 ways, neither one does provide 100% guarantee to meet your needs.
Before going into more detailed descriptions, I must note that Chrome extensions do not provide convenient tools for working on per connection level - sufficiently low level, required for stream capturing. This is by design. This is why the 1-st way is:
To look at other browsers, for example Firefox, which provides low-level APIs for connections. They are already known to be used by similar extensions. You may have a look at MediaStealer. If you do not have a specific requirement to build your system on Chrome, you should possibly move to Firefox.
You can develop a Chrome extension, which intercepts HTTP-requests by means of webRequest API, analyses their headers and extracts media urls (such as containing audio/mpeg MIME-type, for example, in HTTP-headers). Just for a quick example of code you make look at the following SO question - How to change response header in Chrome. Having the url you may force appropriate media download as a file. It will land in default downloads folder and may have unfriendly name. (I made such an extension, but I do not have requirements for further processing). If you need to further process such files, it can be a challenge to monitor them in the folder, and run additional analysis in a separate program.
You may have a look at NPAPI plugins in general, and their streaming APIs in particular. I can imagine that you create a plugin registered for, again, audio/mpeg MIME-type, and receives the data via NPP_NewStream, NPP_WriteReady and NPP_Write methods. The plugin can be wrapped into a Chrome extension. Though I made NPAPI plugins, I never used this API, and I'm not sure it will work as expected. Nethertheless, I'm mentioning this possibility here for completenees. This method requires some coding other than web-coding, meaning C/C++. NB. NPAPI plugins are deprecated and not supported in Chrome since September 2015.
Taking into account that you have some external (to the extension) "fingerprinting service" in mind, which sounds like an intelligent data processing, you may be interested in building all the system out of a browser. For example, you could, possibly, involve a HTTP-proxy, saving media from passing traffic.
If you're writing a Chrome extension, you can use the Chrome tabCapture API to record audio.
chrome.tabCapture.capture({audio: true}, function(stream) {
var recorder = new MediaRecorder(stream);
[...]
});
The rest is left as an exercise to the reader; MDN has more documentation on how to use MediaRecorder.
When this question was asked in 2013, neither chrome.tabCapture nor MediaRecorder existed.
Mac OSX solution using soundflower: http://rogueamoeba.com/freebies/soundflower/
After installing soundflower it should appear as a separate audio device in the sound preferences (apple > system preferences > sound). Divert the computer's audio to the 2ch option (stereo, 16ch is surround), then inside a DAW, such as 'audacity', set the audio input as soundflower. Now the sound should be channeled to your DAW ready for recording.
Note: having diverted the audio from the internal speakers to soundflower you will only be able to hear the audio if the 'soundflowerbed' app is actually open. You know it's open if there's a 8 legged blob in the top right task bar. Clicking this icon gives you the sound flower options.
My privoxy has the following log:
2013-08-28 18:25:27.953 00002f44 Request: api.audioaddict.com/v1/di/listener_sessions.jsonp?_method=POST&callback=_AudioAddict_WP_ListenerSession_create&listener_session%5Bid%5D=null&listener_session%5Bis_premium%5D=false&listener_session%5Bmember_id%5D=null&listener_session%5Bdevice_id%5D=6&listener_session%5Bchannel_id%5D=178&listener_session%5Bstream_set_key%5D=webplayer&_=1377699927926
2013-08-28 18:25:27.969 0000268c Request: api.audioaddict.com/v1/ping.jsonp?callback=_AudioAddict_WP_Ping__ping&_=1377699927928
2013-08-28 18:25:27.985 00002d48 Request: api.audioaddict.com/v1/di/track_history/channel/178.jsonp?callback=_AudioAddict_TrackHistory_Channel&_=1377699927942
2013-08-28 18:25:54.080 00003360 Request: pub7.di.fm/di_progressivepsy_aac?type=.flv
So I got the stream url and record it:
D:\Profiles\user\temp>wget pub7.di.fm/di_progressivepsy_aac?type=.flv
--18:26:32-- http://pub7.di.fm/di_progressivepsy_aac?type=.flv
=> `di_progressivepsy_aac#type=.flv'
Resolving pub7.di.fm... done.
Connecting to pub7.di.fm[67.221.255.50]:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: unspecified [video/x-flv]
[ <=> ] 1,234,151 8.96K/s
I got the file that can be reproduced in any multimedia pleer.
As the title says, if I have an audio node that emits sound and I connect it to two separate GainNodes, which in turn are connected to the Audio Context destination, the sound plays at double speed / double pitch (as if half samples are sent to one gain node and half samples to the other, and the time is halved as well).
I have created an handy jsfiddle here, just drag your sound files in the black rectangle canvas and listen.
// audioContext: Web Audio context
// decoded: decoded audioBuffer
// gainNode1, gainNode2: gain nodes
var bSrc = audioContext.createBufferSource();
bSrc.connect (gainNode1);
bSrc.connect (gainNode2);
gainNode1.connect (audioContext.destination);
gainNode2.connect (audioContext.destination);
bSrc.buffer = decoded;
bSrc.loop = false;
// You'll hear two double-speed buffers playing at unison
bSrc.start(0);
Is that by design? What I would like is to exactly "duplicate" the sound (that will be sent to two different routes, the fiddle is just a proof-of-concept for a bigger project).
Edit:
I tested this on Chrome Version 24.0.1312.56 / Ubuntu 12.10 and the behaviour is present.
The behaviour is also present on Chrome Version 24.0.1312.68 / Ubuntu 12.10
On Chrome Version 24.0.1312.57 / Mac OSX, the Audio API works well and this behaviour is not present.
Could it be a Linux-only issue?
Sounds like a Linux implementation issue. It works for me in Chrome on OS X.