I have 37 cameras in ZoneMinder on protocol RTSP, all writes sound, but I don`t know how write sounds. No sound in streaming and records. Sounds works in VLC.
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How write records in ZoneMinder with sound?
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I understand in a client-server model gRPC can do a bidirectional streaming of data.
I have not tried yet, but want to know will it be possible to stream audio and video data from a source to cloud server using gRPC and then broadcast to multiple client, all in real time ?
TLDR: I would not recommend video over gRPC. Primarily because it wasn't designed for it, so doing it would take a lot of hacking. You should probably take a look at WebRTC + a specific video codec.
More information below:
gRPC has no video compression
When sending video, we want to send things efficiently because sending it raw could require 1GB/s connectivity.
So we use video compression / video encoding. For example, H.264, VP8 or AV1.
Understand how video compression works (eg saving bandwidth by minimising similar data shared between frames in a video)
There is no video encoder for protobufs (the format used by gRPC).
You could then try image compression and save the images in a bytes field (e.g. bytes image_frame = 1;, but this is less efficient and definitely takes up unnecessary space for videos.
It's probably possible to encode frames into protobufs using a video encoder (e.g. H.264) and then decode them to play in applications. However, it might take a lot of hacking/engineering effort. This use case is not what gRPC/protobufs is designed for and not commonly done. Let me know if you hack something together, I would be curious.
gRPC is reliable
gRPC uses TCP (not UDP), which is reliable.
At a glance, reliability might be handy, to avoid corrupting data or lost data. However, depending on the use case (realtime video or audio calls), we may prefer to skip frames if they are dropped or delayed. The losses may be unnoticeable or painless to the user.
If the packet is delayed, it will wait for the packet before playing the rest. (aka. out of order delivery)
If the packet is dropped, it will resend it (aka. packet loss)
Therefore, video conferencing apps usually use WebRTC/RTP (configured to be unreliable)
Having mentioned that, looks like Zoom was able to implement Video-over-WebSockets, which is also a reliable transport (over TCP). So it's not "game-over", just highly not recommended and a lot more effort. They have moved over to WebRTC though.
Data received on the WebSockets goes into a WebAssembly (WASM) based decoder. Audio is fed to an AudioWorklet in browsers that support that. From there the decoded audio is played using the WebAudio “magic” destination node.
I've been doing some research in how to implement a server free, point-to-point video/audio chat (i.e., my own skype without text messaging).
I've been looking for ways to implement it and I had this next ideas:
A multithreaded c++ (cause I know some c++) program getting audio and video (with qt), sending it through 2 different UDP sockets and reading video and audio from 2 other different UDP sockets from the other 'point'. So I'll had to write the UDP server and client multithreaded with a sum of 4 threads: 2 for sending audio and video, other 2 to receive audio and video.
Writing my own protocol to enable video and audio in the same thread, something like parsing half of the packet data size for audio and video buffering, which would leave me with only 2 threads in the application and a lot more 'error prone' code to write.
I've been looking to some real time media protocols, and some of them looked interesting. Maybe study and implement interfaces to this protocols and use them instead of 'creating' my own.
Now, the actual question(s):
Are there some documentation on how to accomplish this? Maybe some 'state of the art' apis/protocols that are being used or well implemented/suited solutions for this problem?
If I choose to implement audio separated from video, is VoIP a possible solution to the audio connection?
Is Qt a good tool for this purpose? I never used Qt before, and for video and audio interfaces I also thought about openframeworks, so I was wondering if anyone has ever used one of this frameworks and if this is the right choice.
I know that my question has no code and that the range of possible answers is wide, but I really need some help here.
Thanks.
First, you should answer on question: How your clients should connect / authorize without server part?
Notes: 1) Skype has servers. 2) A lot of internet users are visiting web throught NAT / Proxy.
Ofc, you can try to implement something for learning proposes, but if you want to create something usefull - try thirdparty solutions that are created by specialists. For example: google libjingle.
You need VOIP library’s :)
There's no need to start from scratch you can use library’s opensource like: opalvoip
I need to record the voice of the student from the client and send it to server to check whether there are any gramatical mistakes in it. I have designed the client in flex.
My problem is, i have no idea of how to store the recorded sound and send it to the server. can you guys give some guidelines? And one more question is red5 better than flash media server?
Red5 and FMS both can do the recording task for you.
For recording, you have to use the NetStream class. That will send the recording real-time to the server. When closing the NetStream make sure the buffer is empty so you don't cut the recording.
You cannot record the sound on the client.the recording and the sending is a single seamless process.
About the servers, I don't know if better is the correct question, it's free so I'd rather use it in this case.
i'm wondering how does the media servers work, do they require large bandwidth if you are doing, let's say, live streaming something like ustream, and there are 10k people watching, do you need a large bandwidth or it is something like p2p ?
I'm more on the client development side with Flash than server admin, but more than likely, yes, you would need a lot of bandwidth to have 10k people watching. The good thing is that with streaming video, you're only downloading the data your watch (unlike progressive). More of an issue would be the number of concurrent connections you could handle per FMS install. 10k would probably require a lot more than 1 server running FMS apps to handle. I'm currently working on a project where we are streaming from 2 installs (beyond the installations of FMS, not sure how they load balanced it) with the hopes of supporting up to something like 2k concurrent connections. I found this article to be pretty helpful (users + bandwidth stats):
http://www.adobe.com/devnet/flashmediaserver/articles/performance_tuning_webcasts.html
The part where "code" meshes with server administration can get pretty daunting (if you ask me)...and every client wants "youtube but with X feature." At 1K a license plus BW, this can get super pricey.
Depending on your needs, you may want to use a 3rd-party FMS company to handle your streaming (especially if it's just for a single event; you can get 'per-event' pricing). Also, I recently used the justin.tv api to create a streaming video feed in Flex. It was pretty painless and all the BW is on them :)
The good part is that once FMS is running, it's super easy to develop with in Actionscript :)
I'm no expert in audio, so if any of you folks are, I'd appreciate your insights on this.
My client has a handful of MP3 podcasts stored at a relatively high bit rate, and I'd like to be able to serve those files to her users at "different" bit rates depending on that user's credentials. (For example, if you're an authenticated user, you might get the full, unaltered stream, but if you're not, you'd get a lower-bit-rate version -- or at least a purposely tweaked lower-quality version than the original.)
Seems like there are two options: downsampling at the source and downsampling at the client. In this case, knowing of course that the source stream would arrive at the client at a high bit rate (and that there are considerations to be made about that, which I realize), I'd prefer to alter the stream at the client somehow, rather than on the server, for several reasons.
Is doing so possible with the Flash Player and ActionScript alone, at runtime (even with a third-party library), or does a scenario like this one require a server-based solution? If the latter, can Flash Media Server handle this requirement specifically? Again, I'd like to avoid using FMS if I can, since she doesn't really have the budget for it, but if that's the only option and it's really an option, I'm open to considering it.
Thanks in advance...
Note: Please don't question the sanity of the request -- I realize it might sound a bit strange, but the requirements are what they are. In that light, for purposes of answering the question, you can ignore the source and delivery path of the bits; all I'm really looking for is an explanation of whether (and ideally how) a Flash client can downsample an MP3 audio stream at runtime, irrespective of whether the audio's arriving over a network connection or being read directly from disk. Thanks much!
I'd prefer to alter the stream at the client somehow, rather than on the server, for several reasons.
Please elucidate the reasons, because resampling on the client end would normally be considered crazy: wasting bandwidth sending the higher-quality version to a user who cannot hear it, and risking a canny user ripping the higher-quality stream at it comes in through the network.
In any case the Flash Player doesn't give you the tools to process audio, only play it.
You shouldn't need FMS to process audio at the server end. You could have a server-side script that loaded the newly-uploaded podcasts and saved them back out as lower-bitrate files which could be served to lowly users via a normal web server. For Python see eg. PyMedia, py-lame; or even a shell script using lame or ffmpeg or something from the command line should be pretty easy to pull off.
If storage is at a premium, have you looked into AAC audio? I believe Flash 9 and 10 on desktop browsers will play it. AAC in my experience takes only half of the size of the comparable MP3 (i.e. a 80kbps AAC will sound the same as a 160kbps MP3).
As for playback quality, if I recall correctly there's audio playback settings in the Publish Settings section in the Flash editor. Wether or not the playback bitrate can be changed at runtime is something I'm not sure of.