I am building a rtmp server, and I want to lag the stream by a minute for my purposes. I was wondering if there was a way to delay the stream for a certain amount of time before processing it.
I am currently using node-media-server for this implementation.
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I am trying to create a RTSP client which live broadcast Audio and Video. I modified the iOS code at link http://www.gdcl.co.uk/downloads.htm and able to broadcast the Video to server properly. But now i am facing issues in broadcasting the audio part. In the link example the code is written in such a way that it writes the Video data to file and than reads the data from the file and upload the NALU's video packets to RTSP server.
For Audio part i am not sure how to proceed on it. Right now what i have tried is that get the audio buffer from mic and than broadcast it to the server directly by adding RTP headers and ALU.. but This approach is not properly working as Audio starts lagging behind and lag increases with time. Can someone let me know if there is some better approach to achieve this and with lip sycn audio/video.
Are you losing any packets on the client? If so, you need to leave "space." If you receive packet 1,2,3,4,6,7, You need to leave space for the missing packet (5).
The other possibility is a what is known as a clock drift problem. The clock (crystal) on your client and server are not perfectly in sync with each other.
This can be caused by environment, temperature changes, etc.
Let's say in a perfect world your server is producing audio samples 20ms audio samples at 48000 hz. Your client is playing them back using a sample rate of 48000 hz. Realistically your client and server are not exactly 48000hz. Your server might be 48000.001 and your client might be 47999.9998. So your server might be delivering faster than your client or vise versa. You would either consume packets too fast and under run the buffer or lag too far behind and overflow the client buffer. In your case, it sounds like the client is playing back too slow and slowly lagging behind the server. You might only lag a couple milliseconds per minute but the issue will keep continuing and it will look like a 1970s lip synced Kung Fu movie.
In other devices, there is often a common clock line to keep things in sync. For example, Video camera clocks, midi clocks. multitrack recorder clocks.
When you deliver data over IP, there is no common clock shared between a client and server. So your issue concerns syncing clocks between disparate devices with no. I have successfully solved this problem using this general approach:
A) Let the client count the rate of packets that come in over a period of time.
B) Let the client count the rate that the packets are consumed (played back).
C) Adjust the sample rate of the client based on A and B.
So your client requires that you adjust the sample rate of the playback. So yes you play it faster or slower. Note that the playback rate change will be very very subtle. You might set the sample rate to be 48000.0001 hz instead of 48000 hz. The difference in pitch would be undetectable by humans as it would only cause a fraction a cent difference in pitch. I gave an explanation of a very simplified approach. There many other nuances and edge cases that must be considered when developing such a control system. You don't just set it and forget it. You need a control system to manage the playback.
An interesting test to demonstrate this is to take two devices with the exact same file. A long recording (say 3 hours) is best. Start them at the same time. After 3 hours of playback, you will notice that one is ahead of the other.
This post explains that it is NOT a trivial task to stream audio and video.
cross posted from dartisans G+ where I got no response so far:
Q: how to do (async) simultaneous stream downloads.
Hi, I'm still learning Dart and, for training, I'd like to create a web page from which I can fetch datas from 1 to 10 URLs that are HTTP streams of binary data. Once I got a chunk of data from each streams , simultaneously, I then perform a computation and proceed to next chunks and so on, ad lib. I need parallelism because the client has much more network bandwith than the servers.
Also I do not want to fully download each URL they're too big to fit in memory or even on local storage. Actually, It's pretty similar to video streaming but it's not video data it's binary data and instead of displaying data I just do some computation and I do that on many streams at a time.
Can I do that with Dart and how ? do dart:io or dart:async have the classes I can use to do that ? do I need to use "webworkers" to spawn 1 to 10 simultaneous http requests ?
any pointers/advices/similar samples would be greatly appreciated.
tl;dr: how to process a HTTP stream of data chunk by chunk and how to parallelize this to process many streams at the same time.
i am trying to make an application that use video/audio streaming through TCP connection, i already done the video streaming with indy10 component(idtcpserver and idtcpclient), is it possible do the same thing but with audio?
Sure.
TCP is just data channel. It is totally agnostic to what kind of data is transferred to it. HTML pages, programs, video, audio - whatever. It is just a data channel within TCP protocol.
However, "streaming" usually means "near to real time". If some frames of video or audio did not arrived during few seconds - they better be skipped and forgotten and newer music or video be played. You would not want your Skype conversation suddenly stuck for a minute and then playback all that minute to you, just because of few seconds network jam. You'd better loose a word or two and then either recover by context or ask the correspondent to repeat. Thus TCP with built-in retransmissions and usually not very large buffers is not a perfect choice for multimedia streaming. Usually UDP + application-implemented integrity control is better choice for it.
I believe you need to use the unit VFW. With avistream, you join video + sound in a compressed stream.
How to serve videos like Youtube does ? Even if the video is long (almost 2 hours long) and is viewed in HD, it would almost instantly play and seeking to not yet loaded parts are very fast.
I'm using a dedicated server from Rackspace with 100Mb up/down for this test, my ping time is below 50ms to the server. My local internet connection is 10Mb, I could maximize my internet connection when I download something from the server so connection to the server is not the issue here.
I'm trying to emulate this and I've tried Real time streaming using Wowza and Pseudostreaming using the H264 Streaming Module. Neither could compare to how fast Youtube delivers video.
Video test file is MP4 (h.264), 300MB, 2 hours long, total bitrate is set to 500kbps, and JWPlayer as the video player
Wowza Streaming (RTMP) - Loading then playing the video is fast, but not as fast as youtube. Seeking is not as fast as well it takes
around 5 - 7 seconds to move to the new position and continue playing the video.
Pseudostreaming H264 Streaming Module (HTTP) - Loading the video takes a long time since its downloading the video header first before
playing it. A 2 hours video has around 2.5MB of MOOV ATOM (video
header file) that it needs to download first before it could play.
Once it starts playing seeking to not downloaded parts is on par with
Wowza but not as fast as Youtube.
What do I need to serve videos with the speed of Youtube? I also need it to buffer/download the video when paused just like Youtube so Real Streaming like Wowza is out.
Pseudostreaming using the H264 Streaming module would have been nice since it does buffer when paused, its just that the initial loading time is very long! Anyway I could remove that initial load time?
What are my other options? I'm open to any other option that I could use in my server.
The way YouTube works is different and they keep on changing the way it works. Doing the reverse engineering on that by capturing the YouTube feeds over wire-shark over last 4 years told me that the pattern is very dynamic. The segmentation is one key, the dual buffer, multiple caching servers and techniques, using the client machine as the buffer render and the functionalities of the player matters a lot. There are many many factors which make YouTube video fast and sleek.
You can emulate the same to some extent but building exactly the same needs loads of efforts and infrastructure.
There is a WWW page with Flash stream on it. I want to download and forward this stream to another streaming server, when possible - replace audio stream (e.g. translate), but without recompressing video stream. Usual way for this ATM is to capture and broadcast Flash player view from the web page, which is obviously suboptimal because video needs to be recompressed, making the quality notably worse and loading the cpu.
Has someone an idea how to do it? VLC seems to be able making relay, but it also seems not to support RTMP at all.
if you're ready to do this programmatically you can use crtmpserver (C++) or red5 (Java) with any RTMP client, otherwise this question doesn't belong to SO