> I've found a reference on what would be the proper behavior, In Unix Network
> Programming, Volume 1: The Sockets Networking API (3rd Edition), it is noted
> that a socket will be ready for writing if the write half of the connection
> is closed, but a write will generate a SIGPIPE. So a chec
On Oct 24, 2013, at 11:35 AM, ssi...@neurosoft.in wrote:
> Any input on this?
Because of this violation of basic email netiquette - i.e., posting the same
question to the mailing list multiple times - future postings from you will be
moderated.
Please don't do this again, otherwise you'll be
On Thu, Oct 24, 2013 at 9:15 PM, Ross Finlayson wrote:
> Thanks for the clarifications, I've looked further and found that send()
> is returning -1, with errno being set to EPIPE. The problem is in "select"
> in BasicTaskScheduler::SingleStep still indicating a writable socket even
> when it is di
I think your problem is here:
void triggerLive555Scheduler(void) {
scheduler->triggerEvent(WAVSource::s_frameReceivedTrigger,
sessionState.source);
}
The problem with this is the second parameter to "triggerEvent()". It needs to
be a pointer to a "WAVSource" object. If you are streami
> Thank you Ross for clarification, its more clear now. Now I am facing issue
> that i have separate thread that pushes audio packets for my device source to
> stream. I trigger event each time I push packet to that queue. I noticed that
> on VLC my audio comes for about a second and then stops.
Any input on this?
On 2013-10-23 11:49, ssi...@neurosoft.in wrote:
Thank you Ross for clarification, its more clear now. Now I am facing
issue that i have separate thread that pushes audio packets for my
device source to stream. I trigger event each time I push packet to
that queue. I noticed th
Hi Ross,
Using the following code I can stop a server media session and restart
it (with different parameters such as multicast address & port). I can
also change from unicast to multicast and back. However when I change
from multicast to unicast the multicast stream continues until I request
Fair enough :-)
I will contact you outside the dev-list to explore options.
thanks,
bob
On Oct 24, 2013 8:58 AM, "Ross Finlayson" wrote:
> The recording files for N > 1 clients work fine with every desktop player
> I use (VLC, Totem, QuickTime, ffplay, Mplayer...etc), however the file no
> lon
Hi Ross,
I have attached
1. my Device source file Wavsource.cpp
2. WaveStreamer .cpp( took a reference from testWavAudioStreamer.cpp) where I
have thread to read the samples and have code for initialization and starting
the session.
Regards
From: Ross Finlayson
Sent: Thursday, October 24, 20
> The recording files for N > 1 clients work fine with every desktop player I
> use (VLC, Totem, QuickTime, ffplay, Mplayer...etc), however the file no
> longer validates as html5 video.
Why don't you try to find out why that is?
In any case, this does not appear to be a problem that I can spen
Ross,
Thanks for your patience and time responding to this issueI do
understand that many of my questions and inquiries are probably outside the
scope of this development list. With that said, I will be brief in my
comments.
With your input and running through numerous permutations from an
im
> I missed the check on "fLimitNumBytesToStream". In this case, should
> "fLimitNumBytesToStream" be initialized to False?
You're right - this is a bug. I've just installed a new version (2013.10.24)
of the code that fixes this. Thanks again for the report.
> Thanks for the clarifications,
> I found the problem that uLawFromPCMAudioSource afterGettingFrame is not
> getting called when I use DeviceSource based design and triggering concept.
> i.e.
> If I am calling FramedSource::afterGetting(this) in doGetNextFrame itself ,
> it is calling afterGettingFrame function in uLawFromPC
Hi Ross,
I found the problem that uLawFromPCMAudioSource afterGettingFrame is not
getting called when I use DeviceSource based design and triggering concept.
i.e.
If I am calling FramedSource::afterGetting(this) in doGetNextFrame itself , it
is calling afterGettingFrame function in uLawFromPC
On Thu, Oct 24, 2013 at 2:51 PM, Ross Finlayson wrote:
> 1. Replies to HTTP GET requests are sometime truncated. As an example,
> curl http://serverip/somets.ts will sometimes result in only part of the
> playlist
>
> I've traced this to fNumBytesToStream is not being initialized when
> create
> I saw the FAQ in live555 website.
> It said the live555 sync the audio and video by RTCP's SR packets.
> So I should create RTCP instance for each RTP source explititly?
No, because (assuming that you are controlling the streaming using RTSP) this
is done implicitly.
(In the RTSP server, this
I saw the FAQ in live555 website.
It said the live555 sync the audio and video by RTCP's SR packets.
So I should create RTCP instance for each RTP source explititly?
At 2013-10-24 05:48:40,"Ross Finlayson" wrote:
So how should I set the duration, then the audio and video would be sync.
Yo
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