Hi I have an mpeg connection that I am trying to read. It has 4 signals. clock, data, valid and sync. Now for some reason I don't see the sync byte when the sync signal is active. I am thinking since there is a sync signal they have eliminated the need for a sync byte. Is there such provision in the MPEG2 protocol. Also since the ISO standard for MPEG2 is paid it would be great if someone can point to some free resources that explain the protocol.
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I have been trying to use WebRTC Data Channel for a game, however, I am unable to consistently send live player data without hitting the queue size limit (8KB) after 50-70 secs of playing.
Sine the data is required to be real-time, I have no use for data that comes out of order. I have initialized the data channel with the following attributes:
negotiated: true,
id: id,
ordered: true,
maxRetransmits: 0,
maxPacketLifetime: 66
The MDN Docs said that the buffer cannot be altered in any way.
Is there anyway I can consistently send data without exceeding the buffer space? I don't mind purging the buffer space as it only contains data that has been clogged up over time.
NOTE: The data is transmitting until the buffer size exceeds the 8KB space.
EDIT: I forgot to add that this issue is only occurring when the two sides are on different networks. When both are within the same LAN, there is no buffering (since higher bandwidth, I presume). I tried to add multiple Data Channels (8 in parallel). However, this only increased the time before the failure occurred again. All 8 buffers were full. I also tried creating a new channel each time the buffer was close to being full and switched to the new DC while closing the previous one that was full, but I found out the hard way (reading Note in MDN Docs) that the buffer space is not released immediately, rather tries to transmit all data in the buffer taking away precious bandwidth.
Thanks in advance.
The maxRetransmits value is ignored if the maxPacketLifetime value is set; thus, you've configured your channel to resend packets for up to 66ms. For your application, it is probably better to use a pure unreliable channel by setting maxPacketLifetime to 0.
As Sean said, there is no way to flush the queue. What you can do is to drop packets before sending them if the channel is congested:
if(dc.bufferedAmount > 0)
return;
dc.send(data);
Finally, you should realise that buffering may happen in the network as well as at the sender: any router can buffer packets when it is congested, and many routers have very large buffers (this is called BufferBloat). The WebRTC stack should prevent you from buffering too much data in the network, but if WebRTC's behaviour is not aggressive enough for your needs, you will need to add explicit feedback from the sender to the receiver in order to avoid having too many packets in flight.
I don't believe you can flush the outbound buffer, you will probably need to watch the bufferedAmount and adjust what you are sending if it grows.
Maybe handle the retransmissions yourselves and discard old data if needed? WebRTC doesn't surface the SACKs from SCTP. So I think you will need to implement something yourself.
It's an interesting problem. Would love to hear the WebRTC W3C WorkGroup takes on it if exposing more info would make things easier for you.
Does Google WebRTC Native implementation has support for SFU?
Does Google WebRTC Native implementation support for integrating custom/hardware encoder/decoder?
Not without alteration.
Internally WebRTC's internal audio/video pipelines are directly tied to encoder/decoders.
PeerConnectionFactory allows you to provide a video decoder/encoder factory, so you can short circuit the logic here, and grab the encoded frames, mock up a stream, and feed them directly into it as a relay, creating a new PeerConnection and setting those streams onto it.
The audio end is more difficult. There isn't a codec factory, so you will have to short circuit the logic there probably by alteration of libwebrtc.
The final question is RTCP termination, and how to override the mechanisms for quality/bandwidth control to not create a "One goes out, they all go out." situation.
Since libwebrtc will be the SFU, it will receive RTCP feedback from its remote peer for the content it is proxying, and vice versa.
For a 1-1 situation, it needs to be able to forward the RTCP feedback to the remote peer.
For multipoint, it needs to perform some logic to determine if one of the peers is problematic, and stop sending it video, switch off its video feed, or attempt to switch to a lower bitrate video stream. Basically it needs to act as a conduit that attempts to predict why/how packet loss is occurring, and keep as many audio/video feeds operating normally at at the highest possible quality for each peer.
How exactly to hijack the RTCP feedback mechanisms in libwebrtc, I think that again will likely require some customization/hooks into libwebrtc
I think it will be easier to try with GStreamer implementation of WebRTC. Although it is still in "Bad Plugins" it is way easier to get or provide encoded audio and video. Actually it is implemented in that in mind - to make implementation of MFU and SFU easier.
I'm developing a USB device driver for a microcontroller (Atmel/Microchip SAMD21, but I think the question is a general one). I need multiple endpoints for control & data, and the USB hardware uses per-endpoint descriptors to (among other things) locate buffers for input and output data.
Since IN data is polled at the host's discretion it makes sense that each endpoint has its own IN buffer, so that any endpoint's data (if it has any to send) is immediately available when polled.
But as far as incoming data from SETUP & OUT transactions is concerned, it occurs to me that I can save memory by configuring all endpoints to use a shared buffer. It seems wasteful for each endpoint to have its own buffer when, given the nature of USB transactions, only one such transaction can occur at a time.
Obviously this approach requires that transaction interrupts are handled sufficiently quickly that the shared buffer is freed and prepared for a new transaction in time for whatever the next transaction might be - but this is already a requirement for the control endpoint, where some SETUP transactions are immediately followed by an OUT.
So, assuming the timing is feasible, is there any other reason why such an approach wouldn't work?
Probably not.
Normally, the USB module on a microcontroller handles OUT packets by keeping track of which packet buffers it has written data to, and it waits for your firmware to say it is done processing the buffer before accepting more data from the computer and overwriting the buffer. If an endpoint has no buffers available to receive more data, but the computer sends an OUT packet to the endpoint, the USB module typically responds to the computer with a NAK packet, which tells the computer it should retry later. In this situation, your firmware can take pretty much as long as it wants to handle the OUT packets.
By having multiple endpoints configured to use the same buffer, you mess up this system. When you receive an OUT packet on any of your endpoints, the USB module would (probably) not know that multiple endpoints use the same buffer, so it would not issue NAK packets on your other OUT endpoints. If it receives another OUT packet right away, it would write it to the same buffer, overwriting the previous packet. Therefore, whenever you receive a packet, your code would have to rush as fast as it can to do something like copying the data out of that buffer, disabling other OUT endpoints, or reassigning buffers.
Even if you can actually get this to work, it means that your scheme to save a little bit of memory turns the servicing of USB events into a real-time task (i.e. a task that requires responses from your code in a few microseconds). If you want to add another real-time task to your system later, it will be very difficult, because you always have to be ready to be interrupted by your USB-handling code.
The SAMD21 has tons of memory (32 KB) so you probably don't need to worry about optimizing this part of it.
I agree with David's Response. You didn't mention the speed of the device you are creating. A low-speed would need just a few 8-byte buffers. A full-speed, a few 64-byte buffers. High-speed, maybe eight 64-byte buffers, depending on your use. A super-speed device, your still only talking a few 512-byte buffers.
I would create a ring buffer for each endpoint. This way you are not moving data around. You are simply using a pointer that points to an entry within a memory ring. A full-speed device with a control endpoint, an interrupt endpoint, and two bulk endpoints, each endpoint having sixteen 64-byte entries per ring, is still only a total of 4k RAM, 1/8th of the total RAM.
However, I am not familiar with the SAMD21, so please check the specification to be sure this will work.
I have a simple UDP streaming protocol that takes RAW H264 video frames and sends them instantly from server side to the client side.
Using this protocol I can get near network RTT latency (no packet resending and I don't care about packet loss), so if I have 20 ms latency from server to the client I can make a video frame to be ready from encoder output to the client side (ready to be decoded) in... let's say 30 ms.
My question is:
Is WebRTC (over UDP) capable of going down to this kind of latencies?
Not taking into account encoding and decoding times, what is the
lowest latency possible I can get with WebRTC for the protocol layer?
I don't know if this kind of latencies will require my own protocol to be more deeply developed or I may go to something more generic like WebRTC for my video server development in order to instantly be supported by every web browser.
WebRTC can have the same low latency as regular SIP/RTP stacks.
WebRTC stack vendors does their best to reduce delay.
For recording and sending out there is no any delay. The stack will send the packets immediately once received from the recorder device and compressed with the selected codec. Some codec's (and some codec settings) might introduce some delay here to enable some features such as FEC.
Regarding the receiver side:
In optimal circumstances the stack should not delay the playback of the packets, so they can be display as soon as they arrive.
However in sub-optimal circumstances (with network delays or packet loss) the stack will introduce a jitter buffer. The lower is the network quality, the higher will be the jitter buffer length.
So, to achieve the lowest delay, you might have to do the followings:
choose a codec with the smallest processing time
remove FEC and disable any other settings which might cause additional delays
remove the jitter buffer (most WebRTC stacks doesn't have a setting for this so you might have to modify the code yourself, but it is an easy modification, because you just need to deactivate a part of the code)
WebRTC uses RTP as the underlying media transport which has only a small additional header at the beginning of the payload compared to plain UDP. This means it should be on par with what you achieve with plain UDP. RTP is heavily used in latency critical environments like real time audio and video (its the media transport in SIP, H.323, XMPP) and thus you can expect the latency to be sufficient for this purpose.
Good evening!
I'm configuring NTP on an embedded Linux system connected with an U-Blox GPS receiver. I've used NTPD and GPSD.
I would like to know what's the technical difference between:
PPS Signal provided by the GPSD shared memory SHM, (Pseudo IP Address 127.127.28.1);
PPS Signal "Stand Alone", but always connected in some way I would like to understand, with GPS (Pseudo IP Address 127.127.22.0)
It is critical for me to understand because I really need an high level synchronization and I would like the right information from the receiver.
Searching all over the web I've found only confused answers to my doubt...
Thanks in advance!
FL
The SHM driver is not designed to provide a PPS signal by itself. So maybe your notion here is misguided.
A PPS signal is used for getting a (precise) notion of the
frequency of the local clock (the one used for measuring external signals), as it just provides a well known timing distance of the "pulses" (1s in this case). Actually pps is a frequency source.
GPSD on the other hand is communicating with some device (could be built into your HW). It then proovides the time data read from the GPS source via shared memory to ntp. This provisioning of data does not guarantee any timing relation (delay). (E.g. could occur earlier or later within the second due to load or scheduling)
From the perspective of ntp, you will have a true date/time label, but you might not know exactly when the related point in time did occur related to your local clock. (Usually not precisely enough for common ntp use cases.) This is where PPS kicks in.
Depending on how the GPS device is being connected to your local machine (parallel, serial, internal bus) you will have some way of getting an interrupt on the pulse from the pps signal. (e.g. with serial connection you usually will get a transition on the DCD pin).
The internal processing of the related interrupt will read the local clock and the resulting timing information is then provided for further processing. This information is exactly what the PPS clock discipline is using and providing to ntp. What you need to configure here, is the offset from the triggering of the pulse to reading the local clock. (Pulse usually is assumed to occur "on the second.)
So, in your configuration, it is likely that the "source" of the PPS signal is the same GPSD is using for providing date/time data (your GPS device).
However, the actual signal used for date/time data and pps is different. Date/time will use a data telegram or some register content read from the GPS device while pps will be a level change on an input pin proveded from this very device.
For details start with the interfacing information from your GPS receiver, especialy any timings stated there. Then look at ntp and figure what driver(s) will allow exploiting such input data for best time quality.