I am attempting to build an interface that allows timing / rhythm (potentially pitch) input to a Web Audio Oscillator node. in effect creating a 'step sequencer'.
What's the best way to trigger scheduled NoteOn for the Web Audio API Oscillator Nodes?
In a specific pattern, i.e. 1/4 notes, 1/8th notes or a user entered pattern.
This is a great question, and in fact I just published an HTML5Rocks article on this very topic: http://www.html5rocks.com/en/tutorials/audio/scheduling/.
Related
I am very confused about the calling sdk specs. They are clear about the fact that only one video stream can be rendered at one time see here...
BUT when I try out the following sample I get video streams for all members of the group call. When I try the other example (both from ms), it behaves like written in the specs... So I am totally confused here why this other example can render more than one video stream in parallel? Can anybody tell me how to understand this? Is it possible or not?
EDIT: I found out that both examples work with multiple videos streams. So it is cool that the service provide more than the specs say, but I do not get the point why the specs tell about that not existing limitations...
Only one video stream is supported on ACS Web (JS) calling SDK, multiple video stream can be rendered for incoming calls but A/V quality is not guaranteed at this stage for more than one video. Support for 4(2x2) and 9(3x3) is on the roadmap and we'll publish support as network bandwidth paired with quality assurance testing and verification is identified and completed.
I am trying to create an on-demand audio streaming platform (similar to Spotify) from scratch. It will have 1000 users (I am optimizing for time to build, not scalability as of right now).
I want to use web-based technologies ( I am experienced with React/Redux/Node). Could I get some advice on the architecture (what technologies I should use for the project)?
Here are things I need help with
What Storage service I should use for my music files (my song catalog is about 50000)
How to stream music from the storage service to each user
What server protocol I should use (RTMP/WebRTC/RTS)
(Optional) How to store data in cache to reduce buffer
I know this is a huge ask so thank you guys for your help in advance
What Storage service I should use for my music files (my song catalog is about 50000)
S3 (or equivalent).
Audio files fit this use case precisely, and you're already using AWS. If you find the cost too high, there are compatible services that are more affordable, all the way down to DIY on Minio.
How to stream music from the storage service to each user
Use a CDN (or multiple CDNs) to optimize delivery and keep the latency low. CDNs are also better at spoon-feeding slow clients.
What server protocol I should use (RTMP/WebRTC/RTS)
Normal HTTP! That's all you need, and that's all that's been necessary for decades for this use case.
RTMP is a dead protocol, only supported by Flash on the client side. Its usage today is limited to sending source streams from video encoders, and even that is well on its way out the door.
WebRTC is intended for low latency connections, like voice calls and video chat. This is not something that matters in a unidirectional stream. You actually want a robust streaming mechanism... not one that drops audio like a cell phone to keep up to realtime.
RTSP is not something you can use in a browser, and is overly complex for what you need.
Just a simple HTTP service is sufficient. Your servers should support ranged requests so that the browser can lose a connection and still pick up right where they left off, without the listener even knowing. (All CDNs support this, as does any properly configured web server.)
(Optional) How to store data in cache to reduce buffer
CDNs will generally improve performance of the initial connect and load. I also recommend pre-loading the next track to be played in the list so that you can start it immediately. In most browsers, you can actually start the next track at the tail end of the previous track for a smooth transition.
I'm working on an IoT project that involves a sensor transmitting its values to an IoT platform. One of the platforms that I'm currently testing is Thingsboard, it is Open Source and I find it quite easy to manage.
My sensor is transmitting active energy indexes to Thingsboard. Using these values, I would like to calculate and show on a widget the values of the active power (= k*[ActiveEnergy(n)- ActiveEnergy(n-1)/Time(n)-Time(n-1)]). This basically means that I want to have access to history data, use this data to generate new data and inject it to my device.
Thingsboard uses Cassandra database to save history values.
One alternative to my problem could be to find a way to communicate with the database via a Web API for example, do the processing and send back the active power by MQTT or HTTP on my device using its access token.
Is this possible?
Is there a better alternative to my problem?
There are several options how to achieve this (based on a layer or component of the system):
1) Visualization layer only. Probably the most simple one. There is an option to apply post-processing function. The function has following signature:
function(time, value, prevValue)
Please note that prevTime is missing, but we may add this in future releases.
post processing function
2) Data processing layer. Use advanced analytics frameworks like Apache Spark to post-process your data using sliding time window, for example.
See our integration article about this.
I am planning to write a nodeJS server for streaming videos, One of my critical requirement is
to prevent video download( as much as possible ), something similar to safaribooksonline.com
I am planning to use amazon s3 for storage and nodeJS for streaming the videos to the client.
I want to know if nodeJS is the right tool for streaming videos( max size 100mb ) for an application expecting lot of users. If not then what are the alternatives ?
Let me know if any additional details are required.
In very simple terms you can't prevent video download. If a rogue client wants to do it they they generally can - the video has to make it to the client for the client to be able to play it back.
What is most commonly done is to encrypt the video so the downloaded version is unplayable without the right decryption key. A DRM system will allow the client play the video, without being able to copy it (depending on how determined the user is - a high quality camera pointed at a high quality screen is hard to protect against (!). In these scenarios, other tracing technologies come in to play).
As others have mentioned in the comments, streaming servers are not simple - they have to handle a wide range or encoders, packaging formats, streaming formats etc to allow as much each as possible and will have quite complicated mechanisms to ensure speed and reduce file storage requirements.
It might be an idea to look at some open source streaming servers to get a feel for the area, for example:
VideoLan (http://www.videolan.org/vlc/streaming.html)
GStreamer (https://gstreamer.freedesktop.org)
You can still use noedejs for the main web server component of your solution and just hand off the video streaming to the specialised streaming engines, if this meets your needs.
I have a telephony scenario in which the following happens:
Customer calls a Voice Gateway
TCL script runs and a code is taken from customer
Authentication is done through a RADIUS server
Customer will hear correct voice menu
The problem is that RADIUS server must connect to a SQL Database and check the credentials. I have currently designed the solution using cisco secure ACS and through managed stored procedures on MS SQL server.
My question is: Is the VoiceXML a better tool to do this job and because some extenstions and wrappers of VoiceXML exists in .net, does it fit in this simple scenario??
Sincerely speaking, I am a little confisued with the technology and looking for a good tutorial on its features as well.
Thanks
In a strict sense, only step 4 is implemented by VoiceXML. Other aspects are handled by the platform or external code. VoiceXML is the standards mechanism for implementing step 4, but if all you are going to do is limited audio output and simple input, it may be overkill depending on the solutions available to you.
The following is just an example of a way to solve your problem and is fairly fictitious given I don't know anything about your environment nor constraints.
Given most VoiceXML platforms, upon receiving of a call your VoiceXML application will be executed. If this is a servlet/ASP based solution, you can perform steps 2 & 3 then generate/return the VoiceXML to play the menu, gather the input and move to the next step. If this is a static VoiceXML 2.1 solution, you can use a Data element call to make an HTTP request to a system that can perform these actions. The system will need to return XML that the Javascript/ECMAScript in VoiceXML application can parse and provide the correct audio output and input processing.
Since you are asking about VoiceXML, I'm assuming your challenge is the telephony aspect of the problem. Unless you have a system already available, choosing and activating a premise or hosted solution is far more complicated than the call flow code involved. Depending on your requirements, there are solutions as low as a single line, analog modem that supports audio output and DTMF input to massively scaled on premise and hosted solutions to handle 10,000s of concurrent calls that implement VoiceXML as well as a wide range of other call flow technologies.
VoiceXML would work fine in this scenario. There is a an open source project called VoiceModel that uses ASP.NET MVC to generate the VoiceXML and therefore integrates nicely with the .NET stack. There are a lot of examples in the project with discussions on how to use the examples in this blog. The examples use Voxeo Prophecy as the VoiceXML platform which has a SIP interface that will connect with a Voice Gateway. You can download two ports for free to try it out.