[Live-devel] Re: Audio Video Synchronization-some queries
Luca Piccarreta
piccarre at elet.polimi.it
Thu Sep 1 18:53:52 PDT 2005
>From what I understand, RTP timestamps are
"sampling timestamps" + random offset.
Therefore they are a very good timing source.
(Even after a Pause/Play command, I suppose)
But NTP timestamps are still needed
- to correctly synchronize streams with the
same clock source (SSRC), which might actually come
from different inputs (it is in my experience typical
that those "random offsets" in RTP timestamps are
different even in the simplest case A/V RTSP+UDP
streaming).
- to avoid drifts between client and server clocks.
(it might happen that the clients cache lots of data
because the server clock is running too fast!)
Just an opinion, as I'm really a newbie!
Luca.
----- Original Message -----
From: "shobana meenakshi" <vshobs at yahoo.com>
To: <live-devel at ns.live.com>
Sent: Thursday, September 01, 2005 5:17 PM
Subject: [Live-devel] Re: Audio Video Synchronization-some queries
Hi all
As I am new to implementing RTSP/RTCP protocol, I thought I would seek help
from this mailing list.
I would like to explain my understanding here with respect to audio video
synchronization after PAUSE/RESUME and during seek.Request you to correct me
if Iam wrong.There are ideally 2 ways the streaming server gives the
client,synchronization information for audio and video.
1. RTSP PLAY Response
If RTSP is used as the control channel, after an RTSP PAUSE and PLAY the
server sends a play response before sending the first packet after resume in
which it tells the RTP time stamp for both audio and video which would be in
synch.
Ideally the implementation on the Client for the audio video synchronization
would be , after getting the first packet after resume calculate the offset
for the first packet's RTP timestamp from the reference RTPtimestamp got
from the PLay response in both the audio and video threads seperately and
increment the presentation time by that offset(after dividing it by the RTp
timestamp frequency say 44100 for audio) so that they would be in synch.
Am I right in this understanding?
Now given this understanding is right, we won't need the wallclock
synchronization at all as the RTP timestamp is purely based on the number of
samples sent in the packet in which case we won't need the RTCP
implementation at all.
What am I missing here?
2.RTCP sender report
The RTP sender report sends the NTP timestamp which would be synchronized
for both server and client and for both audio and video and an RTP timestamp
associated with it.Ideally after pause and resume after we get the RTCP
sender report, we take the RTP timesatamp sent in the RTCP sender report use
that as a reference calculate the offset of the first audio and video
packets after resume from the reference RTP timestamp and calculate the
presentation time in the same way as explained above.
If this is correct what I don't understand is
1.How is the NTP time associated with the RTP timestamp if RTp timestamp is
purely based on the sample count in the packet?
In general I have the following queries regarding how the server sends the
RTP time stamp in the play response as well as in the RTCP sender report.
1.Suppose the RTPtimestamp for audio=12345 and RTP Timestamp for
Video=123456 at the point they were paused.
After Resume the first audio and video packet should have an RTP timestamp
of say audio=12360 and video=123486 ideally if we assume that the packets
sent has 15 audio samples and 30 video samples respectively.Is my
understanding right or is the duration of PAUSE has to do anything with the
RTPtimestamps(in the sense would the server calculate the RTPtimestamp of
the first packet based on what it would have been if we did not pause at all
based on the wall clocktime?)
It would be really helpful if anyone could validate or clarify my
understanding.Though this is slightly out of topic I think it would be
really useful for those who are new to this.
Thanks and Regards
Shobana V
live-devel-request at ns.live.com wrote:
Send live-devel mailing list submissions to
live-devel at lists.live.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.live.com/mailman/listinfo/live-devel
or, via email, send a message with subject or body 'help' to
live-devel-request at lists.live.com
You can reach the person managing the list at
live-devel-owner at lists.live.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of live-devel digest..."
Today's Topics:
1. RTP URL (Omenkeukwu, Gregory)
2. Re: RTP URL (Ross Finlayson)
3. RTSP url (Uday Sinha)
4. Re: RTSP url (Ross Finlayson)
5. Re: RTSP url (Uday Sinha)
6. Re: RTSP url (Ross Finlayson)
7. Re: RTSP url (Uday Sinha)
8. Re: RTSP url (Ross Finlayson)
9. WebCam as Frame Source (Ioannis Skazikis)
----------------------------------------------------------------------
Message: 1
Date: Wed, 31 Aug 2005 16:14:06 -0500
From: "Omenkeukwu, Gregory"
Subject: [Live-devel] RTP URL
To: "'live-devel at lists.live.com'"
Message-ID:
<3D083A0287FC9648B9D6F907A94A354D01EA8342 at xchsrv01.DEERFIELD.AIM.local>
Content-Type: text/plain; charset=iso-8859-1
>
>
> Hello,
> I am developing a web based application for RTP urls but I am having
> problems finding internet radio stations transmitting RTP. It seems most
> of the internet radio stations are transmitting HTTP. Please I will like
> to get a list of online streaming audio urls in RTP. If you have any
> available please send them to me, I will greatly appreciate it. Thanks
>
> Greg Omenkeukwu
> gregsnips at yahoo.com
------------------------------
Message: 2
Date: Wed, 31 Aug 2005 14:29:16 -0700
From: Ross Finlayson
Subject: Re: [Live-devel] RTP URL
To: "LIVE.COM Streaming Media - development & use"
Message-ID: <6.2.3.4.1.20050831142550.03209eb0 at localhost>
Content-Type: text/plain; charset="us-ascii"; format=flowed
> > I am developing a web based application for RTP urls but I am having
> > problems finding internet radio stations transmitting RTP. It seems most
> > of the internet radio stations are transmitting HTTP. Please I will like
> > to get a list of online streaming audio urls in RTP. If you have any
> > available please send them to me, I will greatly appreciate it. Thanks
RTP URLs have never been standardized, and are generally a bad idea,
because they don't convey enough information to fully describe a stream.
Instead, RTSP URLs (i.e., beginning with "rtsp://") are the standard.
You can find several such URLs by typing "url:rtsp" into Google.
Ross Finlayson
LIVE.COM
------------------------------
Message: 3
Date: Wed, 31 Aug 2005 22:07:15 +0000
From: "Uday Sinha"
Subject: [Live-devel] RTSP url
To: live-devel at ns.live.com
Message-ID:
Content-Type: text/plain; charset="us-ascii"
An HTML attachment was scrubbed...
URL:
http://lists.live.com/pipermail/live-devel/attachments/20050831/569423a2/attachment-0001.html
------------------------------
Message: 4
Date: Wed, 31 Aug 2005 15:18:39 -0700
From: Ross Finlayson
Subject: Re: [Live-devel] RTSP url
To: "LIVE.COM Streaming Media - development & use"
Message-ID: <6.2.3.4.1.20050831151746.03200430 at localhost>
Content-Type: text/plain; charset="us-ascii"; format=flowed
>I am trying use LIVE.COM streaming client using openRTSP.exe. I need
>a rtsp url, preferably with windows, mpeg content, and that does not
>use real codecs since they are not supported on LIVE.COM's. Will
>appreciate a url that works with LIVE.COM streamer. Thanks.
Try
rtsp://christianchallenge.org/ccistream/sunday.mp4
This is MPEG-4 video and audio. It plays in QuickTime Player, VLC
(which uses the "LIVE.COM Streaming Media" code), and "openRTSP".
Ross Finlayson
LIVE.COM
------------------------------
Message: 5
Date: Wed, 31 Aug 2005 23:19:52 +0000
From: "Uday Sinha"
Subject: Re: [Live-devel] RTSP url
To: live-devel at ns.live.com
Message-ID:
Content-Type: text/plain; charset="us-ascii"
An HTML attachment was scrubbed...
URL:
http://lists.live.com/pipermail/live-devel/attachments/20050831/33dc2a13/attachment-0001.html
------------------------------
Message: 6
Date: Wed, 31 Aug 2005 16:26:49 -0700
From: Ross Finlayson
Subject: Re: [Live-devel] RTSP url
To: "LIVE.COM Streaming Media - development & use"
Message-ID: <6.2.3.4.1.20050831162510.03213720 at localhost>
Content-Type: text/plain; charset="us-ascii"; format=flowed
At 04:19 PM 8/31/2005, you wrote:
>Thanks for RTSP url. However, I do not get the audio on speaker. An
>audio file [audio-MPEG4-GENERIC-1] is created yet the file is empty.
>Am I doing something wrong.
Yes, the "-a" option requests that audio be output to *standard
output* (and that video not be received).
If you omit the "-a" option, you'll get two output files - one
containing video; the other containing audio.
Ross Finlayson
LIVE.COM
------------------------------
Message: 7
Date: Thu, 01 Sep 2005 00:38:04 +0000
From: "Uday Sinha"
Subject: Re: [Live-devel] RTSP url
To: live-devel at ns.live.com
Message-ID:
Content-Type: text/plain; charset="us-ascii"
An HTML attachment was scrubbed...
URL:
http://lists.live.com/pipermail/live-devel/attachments/20050901/fb2c10a9/attachment-0001.html
------------------------------
Message: 8
Date: Wed, 31 Aug 2005 17:44:12 -0700
From: Ross Finlayson
Subject: Re: [Live-devel] RTSP url
To: "LIVE.COM Streaming Media - development & use"
Message-ID: <6.2.3.4.1.20050831174259.03228110 at localhost>
Content-Type: text/plain; charset="us-ascii"; format=flowed
At 05:38 PM 8/31/2005, you wrote:
>Looks like there is some problem with the server. Can you stream on
>your end with following command:
>
>openRTSP -e 50.0 rtsp://christianchallenge.org/ccistream/sunday.mp4
Yes, it works for me.
Perhaps you're yet another person who's behind a firewall that's
blocking UDP? If so, add the "-t" option to "openRTSP", to request
RTP-over-TCP streaming.
Ross Finlayson
LIVE.COM
------------------------------
Message: 9
Date: Thu, 01 Sep 2005 08:32:50 +0300
From: Ioannis Skazikis
Subject: [Live-devel] WebCam as Frame Source
To: live-devel at ns.live.com
Message-ID: <43169282.9080004 at intracom.gr>
Content-Type: text/plain; charset="iso-8859-1"
Hallo,
I try to build an RTSP Live Video Stream server based on the live media
library.
I have also read the live Media FAQ + Mailing List and tryed to follow
your answers to questions from the mailing list archive from others
having the same problem like me.
The program is based on the testMPEG4VideoStreammer testprogramm found
in the liveMedia library.
Based on the DeviseSource.cpp I wrote an DeviceSourceCammera subclass of
the FramedSource" which opens a USB Camera, grubs an Image and decodes
it with help of the Revel Lib (XVID encoding Lib) to MPEG4.
The functions which open the device and starts the Server look like this:
void init_play(){
// Open the input source
DeviceParameters params;
fileSource = DeviceSourceCammera::createNew(*env, params);
if (fileSource == NULL) {
*env << "Unable to open source\n";
exit(1);
}
}
void play() {
FramedSource* videoES = fileSource;
// Create a framer for the Video Elementary Stream:
videoSource = MPEG4VideoStreamFramer::createNew(*env, videoES);
// Finally, start playing:
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}
>From debuging outputs implemented in the DeviceSourceCammera.cpp by
running the programm I can see that it really opens the device, grab a
frame and encodes it (the procedure repeats it self), but if I start the
mplayer, it connects to the servers reads the sdp , it writes as output
at the end "Found video stream: 0" ,but it does not starts to play the
stream. It stays there with out do nothing (like waiting frames, but
never receives them)
Any Idea why (or what?) is happening?
The DeviceSourceCammera.cpp is included in this e-mail.
lvsMain.cpp is the "Main" file with the server start up code (also
included in this e-mail)
DeviceUtil.hh is the Header file which includes the basic Function of my
code for handling the WebCam device. (this works, I have encoded a
video file, before to play with the LiveMedia Lib.)
The subclass Functions look like this. If you need more details please
ask me or have a look in the attached files...
DeviceSourceCammera::DeviceSourceCammera(UsageEnvironment& env,
DeviceParameters params)
: DeviceSource(env, params){
init(); // Open Device
}
void DeviceSourceCammera::doGetNextFrame() {
grab_image(); //Take a Frame from the Camera
if (0 /* the source stops being readable */) {
handleClosure(this);
fprintf(stderr,"In Grab Image V4l, the source stops being readable
!!!!\n");
return;
}
deliverFrame(); // call DeviceSourceCammera::deliverFrame() Function.
}
DeviceSourceCammera::deliverFrame() {
//Encoding the grub frame from the Camera
if (get_grab_size() > fMaxSize){
frameSize = fMaxSize;
}
else frameSize = get_grab_size();
fTo = (unsigned char*)malloc(frameSize);
memcpy(fTo, frame.pixels, frameSize); // frame.pixels is the buffer
where the grubed frame is now encoded, framesize the size of the encoded
frame
}
Thank you very much for your help and your time !!!!
Sorry about my bad english too.
Ioannis.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: DeviceSourceCammera.cpp
Type: text/x-c++src
Size: 6173 bytes
Desc: not available
Url :
http://lists.live.com/pipermail/live-devel/attachments/20050901/95c419c3/DeviceSourceCammera.bin
-------------- next part --------------
A non-text attachment was scrubbed...
Name: DeviceSourceCammera.hh
Type: text/x-c++hdr
Size: 679 bytes
Desc: not available
Url :
http://lists.live.com/pipermail/live-devel/attachments/20050901/95c419c3/DeviceSourceCammera-0001.bin
-------------- next part --------------
A non-text attachment was scrubbed...
Name: DeviceUtil.cpp
Type: text/x-c++src
Size: 3680 bytes
Desc: not available
Url :
http://lists.live.com/pipermail/live-devel/attachments/20050901/95c419c3/DeviceUtil.bin
-------------- next part --------------
A non-text attachment was scrubbed...
Name: DeviceUtil.hh
Type: text/x-c++hdr
Size: 501 bytes
Desc: not available
Url :
http://lists.live.com/pipermail/live-devel/attachments/20050901/95c419c3/DeviceUtil-0001.bin
-------------- next part --------------
A non-text attachment was scrubbed...
Name: lvsMain.cpp
Type: text/x-c++src
Size: 3670 bytes
Desc: not available
Url :
http://lists.live.com/pipermail/live-devel/attachments/20050901/95c419c3/lvsMain.bin
-------------- next part --------------
A non-text attachment was scrubbed...
Name: revel.h
Type: text/x-chdr
Size: 9981 bytes
Desc: not available
Url :
http://lists.live.com/pipermail/live-devel/attachments/20050901/95c419c3/revel.bin
------------------------------
_______________________________________________
live-devel mailing list
live-devel at lists.live.com
http://lists.live.com/mailman/listinfo/live-devel
End of live-devel Digest, Vol 22, Issue 34
******************************************
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
--------------------------------------------------------------------------------
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live.com
> http://lists.live.com/mailman/listinfo/live-devel
>
More information about the live-devel
mailing list