Posted: Wed Feb 28, 2007 5:46 pm Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi all
We have finally got the amr audio working with app_mp4 and the h324m
library.
The amr track can be encoded using ffmpeg (downloading the amr reference
codec)
and hinting it with mpeg4ip.
The audio recording is still a little buggy but i hope to get it working
really soon.
If you find any problem with any of the apps or libraries let me know and
I'll take a look
at it.
Posted: Thu Mar 01, 2007 10:44 am Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi,
On Wed, 2007-02-28 at 18:32 +0100, Sergio Garcia Murillo wrote:
Quote:
Hi all
Quote:
If you find any problem with any of the apps or libraries let me know and
I'll take a look
at it.
I'm trying to update everything in order to test audio,
but... do you remember about TIFFReverseBits in h324m.cpp ?
I have to comment it out because of inverted bits from misdn...
but now, as soon as I place a call, I get from asterisk cli:
asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_h324m.so:
undefined symbol: TIFFReverseBits
And then * crashes.
uhm... maybe something must be changed also into app_h324m ?
greetings,
Matteo.
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
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Posted: Thu Mar 01, 2007 10:52 am Post subject: [Asterisk-video] H324M AMR audio working!!!
From: "matteo brancaleoni" <mbrancaleoni@espia.it>
Sent: Thursday, March 01, 2007 11:42 AM
Quote:
Hi,
On Wed, 2007-02-28 at 18:32 +0100, Sergio Garcia Murillo wrote:
> Hi all
> If you find any problem with any of the apps or libraries let me know
and
Quote:
> I'll take a look
> at it.
I'm trying to update everything in order to test audio,
but... do you remember about TIFFReverseBits in h324m.cpp ?
I have to comment it out because of inverted bits from misdn...
but now, as soon as I place a call, I get from asterisk cli:
But did you comment the whole function or just the function call in the
h324m.cpp???
Quote:
asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_h324m.so:
undefined symbol: TIFFReverseBits
Make clean on both library an app and try objdump -T libh324m.so to
see if the symbol is defined correctly in the library or not.
Posted: Thu Mar 01, 2007 10:55 am Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi,
On Thu, 2007-03-01 at 12:04 +0100, Sergio Garcia Murillo wrote:
Quote:
>
> I'm trying to update everything in order to test audio,
> but... do you remember about TIFFReverseBits in h324m.cpp ?
> I have to comment it out because of inverted bits from misdn...
> but now, as soon as I place a call, I get from asterisk cli:
But did you comment the whole function or just the function call in the
h324m.cpp???
only the function call in the read & write routines...
exactly as did before :)
Quote:
> asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_h324m.so:
> undefined symbol: TIFFReverseBits
Make clean on both library an app and try objdump -T libh324m.so to
see if the symbol is defined correctly in the library or not.
Ok, I'll do and report back.
matteo.
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
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Posted: Thu Mar 01, 2007 11:47 am Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi,
On Thu, 2007-03-01 at 12:04 +0100, Sergio Garcia Murillo wrote:
Quote:
> asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_h324m.so:
> undefined symbol: TIFFReverseBits
Make clean on both library an app and try objdump -T libh324m.so to
see if the symbol is defined correctly in the library or not.
For being 100% sure I checked out from svn again.
If I compile the lib as-is:
[root@vmart-beta libh324m]# !obj
objdump -T libh324m.so | grep -i TIFFReverseBits
000d61a4 g DF .text 000000ae Base TIFFReverseBits
If I comment out in h324m.cpp in this way:
int H324MSessionRead(void * id,unsigned char *buffer,int len)
{
//TIFFReverseBits(buffer,len);
return ((H324MSession*)id)->Read(buffer,len);
}
int H324MSessionWrite(void * id,unsigned char *buffer,int len)
{
int ret = ((H324MSession*)id)->Write(buffer,len);
//TIFFReverseBits(buffer,len);
return ret;
}
I have no more output from objdump...
Hints ?
Matteo
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
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Posted: Thu Mar 01, 2007 12:20 pm Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi,
On Thu, 2007-03-01 at 12:47 +0100, matteo brancaleoni wrote:
Quote:
Hi,
On Thu, 2007-03-01 at 12:04 +0100, Sergio Garcia Murillo wrote:
> > asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_h324m.so:
> > undefined symbol: TIFFReverseBits
>
> Make clean on both library an app and try objdump -T libh324m.so to
> see if the symbol is defined correctly in the library or not.
For being 100% sure I checked out from svn again.
I did a quick&dirty hack, like defining a new unused function
that calls TIFFReverseBits and does nothing at all.
In this way the object gets loaded and everything works :)
Audio also :)
Matteo.
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
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Posted: Thu Mar 01, 2007 1:40 pm Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi,
On Thu, 2007-03-01 at 13:46 +0100, Sergio Garcia Murillo wrote:
Quote:
From: "matteo brancaleoni" <mbrancaleoni@espia.it>
Sent: Thursday, March 01, 2007 1:06 PM
> > > > asterisk: symbol lookup error:
/usr/lib/asterisk/modules/app_h324m.so:
> > > > undefined symbol: TIFFReverseBits
I have updated a new verson that should solve the problem also.
Please try it id you can to make sure I haven't broke anything..
Yes, seems ok :)
thanks a lot :)
btw... the audio is a bit "broken", I mean that
has some interruptions in it.
But maybe is the encoding... I'll check.
When switching between 2 videos, the audio always
arrives immediately and the video a bit later.
But maybe this 3G related :)
really good work.
Now we need only an amr codec and we can bridge sip :)
Matteo
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
_______________________________________________
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Posted: Thu Mar 01, 2007 2:49 pm Post subject: [Asterisk-video] H324M AMR audio working!!!
The problem with bridging to SIP is bigger than you think
My H324m stack is directly implemented into chan_zap, so I can place 3G call with dial(zap/gv1/number) or I receive video call from a cellphone in H263 mode.
Also since I wrote codec_amr and format_amr, i gave chan_zap the amr capability so the call is transcoded.
Currently when I bridge to SIP it works but there is a BIG problem, it's the FPS that the softphone is sending.
Basically, if the H263 video coded by the softphone is about 50kb/s its fine. but if it's bigger (which is usually the case) then you loose frame.
Also, the softphone is not usually capable of sending a SIP info as Video Fast Update asking the softphone to reencode an I-frame.
That's why I'm currenty working on the transcoding of H263 <> H263.
--
Amin Ramtin
> Subject: Re: [Asterisk-video] H324M AMR audio working!!!
On Thu, 2007-03-01 at 13:46 +0100, Sergio Garcia Murillo wrote:
> From: "matteo brancaleoni" <mbrancaleoni@espia.it>
> Sent: Thursday, March 01, 2007 1:06 PM
> > > > > asterisk: symbol lookup error:
> /usr/lib/asterisk/modules/app_h324m.so:
> > > > > undefined symbol: TIFFReverseBits
> I have updated a new verson that should solve the problem also.
> Please try it id you can to make sure I haven't broke anything..
Yes, seems ok :)
thanks a lot :)
btw... the audio is a bit "broken", I mean that
has some interruptions in it.
But maybe is the encoding... I'll check.
When switching between 2 videos, the audio always
arrives immediately and the video a bit later.
But maybe this 3G related :)
really good work.
Now we need only an amr codec and we can bridge sip :)
Matteo
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
_______________________________________________
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Posted: Thu Mar 01, 2007 3:08 pm Post subject: [Asterisk-video] H324M AMR audio working!!!
It's normal, Amin: for 3G calls you can't exceed 51200 bit/s for the video stream and 12200 bit/s for the audio stream, as stated in the H324M draft(s).
So it will be better to keep both streams a little bit under the maximum allowed to be safe.
Cesare
At 15.48 01/03/2007, you wrote:
Quote:
The problem with bridging to SIP is bigger than you think
My H324m stack is directly implemented into chan_zap, so I can place 3G call with dial(zap/gv1/number) or I receive video call from a cellphone in H263 mode.
Also since I wrote codec_amr and format_amr, i gave chan_zap the amr capability so the call is transcoded.
Currently when I bridge to SIP it works but there is a BIG problem, it's the FPS that the softphone is sending.
Basically, if the H263 video coded by the softphone is about 50kb/s its fine. but if it's bigger (which is usually the case) then you loose frame.
Also, the softphone is not usually capable of sending a SIP info as Video Fast Update asking the softphone to reencode an I-frame.
That's why I'm currenty working on the transcoding of H263 <> H263.
--
Amin Ramtin
> Subject: Re: [Asterisk-video] H324M AMR audio working!!!
> From: mbrancaleoni@espia.it
> To: asterisk-video@lists.digium.com
> Date: Thu, 1 Mar 2007 14:39:23 +0100
>
> Hi,
>
> On Thu, 2007-03-01 at 13:46 +0100, Sergio Garcia Murillo wrote:
> > From: "matteo brancaleoni" <mbrancaleoni@espia.it>
> > Sent: Thursday, March 01, 2007 1:06 PM
> > > > > > asterisk: symbol lookup error:
> > /usr/lib/asterisk/modules/app_h324m.so:
> > > > > > undefined symbol: TIFFReverseBits
> > I have updated a new verson that should solve the problem also.
> > Please try it id you can to make sure I haven't broke anything..
>
> Yes, seems ok :)
> thanks a lot :)
>
> btw... the audio is a bit "broken", I mean that
> has some interruptions in it.
> But maybe is the encoding... I'll check.
>
> When switching between 2 videos, the audio always
> arrives immediately and the video a bit later.
> But maybe this 3G related :)
>
> really good work.
> Now we need only an amr codec and we can bridge sip :)
>
> Matteo
>
> --
> Matteo Brancaleoni
> R&D Director
> Tel :+39.02.70633354
> Voip :sip:matteo@sip.voismart.it
>
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Posted: Thu Mar 01, 2007 3:14 pm Post subject: [Asterisk-video] H324M AMR audio working!!!
Basically if the softphone is sending H263 way Higher than 50kb, i have to transcode ! and that was my point.
So i was currently doing this job with ffmpeg...
It's normal, Amin: for 3G calls you can't exceed 51200 bit/s for the video stream and 12200 bit/s for the audio stream, as stated in the H324M draft(s).
So it will be better to keep both streams a little bit under the maximum allowed to be safe.
Cesare
At 15.48 01/03/2007, you wrote:
Quote:
The problem with bridging to SIP is bigger than you think
My H324m stack is directly implemented into chan_zap, so I can place 3G call with dial(zap/gv1/number) or I receive video call from a cellphone in H263 mode.
Also since I wrote codec_amr and format_amr, i gave chan_zap the amr capability so the call is transcoded.
Currently when I bridge to SIP it works but there is a BIG problem, it's the FPS that the softphone is sending.
Basically, if the H263 video coded by the softphone is about 50kb/s its fine. but if it's bigger (which is usually the case) then you loose frame.
Also, the softphone is not usually capable of sending a SIP info as Video Fast Update asking the softphone to reencode an I-frame.
That's why I'm currenty working on the transcoding of H263 <> H263.
On Thu, 2007-03-01 at 13:46 +0100, Sergio Garcia Murillo wrote:
> From: "matteo brancaleoni" <mbrancaleoni@espia.it>
> Sent: Thursday, March 01, 2007 1:06 PM
> > > > > asterisk: symbol lookup error:
> /usr/lib/asterisk/modules/app_h324m.so:
> > > > > undefined symbol: TIFFReverseBits
> I have updated a new verson that should solve the problem also.
> Please try it id you can to make sure I haven't broke anything..
Yes, seems ok :)
thanks a lot :)
btw... the audio is a bit "broken", I mean that
has some interruptions in it.
But maybe is the encoding... I'll check.
When switching between 2 videos, the audio always
arrives immediately and the video a bit later.
But maybe this 3G related :)
really good work.
Now we need only an amr codec and we can bridge sip :)
Matteo
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
_______________________________________________
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Hi,
btw... the audio is a bit "broken", I mean that
has some interruptions in it.
But maybe is the encoding... I'll check.
You can check it using the if2amr test program with the out media dump file
and see if
it also has the interruptions, it probably won't so the problem is mor
probable to be with
the video problem
Quote:
When switching between 2 videos, the audio always
arrives immediately and the video a bit later.
But maybe this 3G related :)
No, the problem is a bandwith one, and related with ffmpeg (and it's also
proably affecting
audio also)
The main problem using ffmpeg is that it's a vbr encoding, and althought you
can adjust
adjust a lot of parameters I still have not been able to find the way of
make it behave
correctly.
The problem is the length of the I-Frames, that is to long ( you can check
it encoding with -vstat).
It can returns frames of 10Kb long , and the rest of P-Frames is just a few
bytes. With a file
playing in a computer there is not much problem or even streaming it using
som buffer.
But we are sending it through an isdn channel so it's going to take a few
seconds just to
send the first I frame (you can calculate it) and that's probably the delay
you're seeing in
the video..
The audio could be affected as well because in the meanwhile all those
frames are been queued
in the stack and rigth now i don't priorize them, so its probably that the
amr channel is starved
and the audio frames are not sent with the correct timing...
Quote:
really good work.
Now we need only an amr codec and we can bridge sip :)
Thanxs, It won't be too diffciult using the libavocodec from ffpmeg..
Posted: Thu Mar 01, 2007 3:49 pm Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi Ratmin, it's good to see you're alive and well...
Quote:
From: Ramtin Amin
Sent: Thursday, March 01, 2007 3:48 PM
The problem with bridging to SIP is bigger than you think
My H324m stack is directly implemented into chan_zap, so I can place 3G
call with dial(zap/gv1/number) or I receive video call from a cellphone in
H263 mode.
Good, and what it's the difference bridging the call to a sip channel?
Beside of that sepparating the stack from zap allows the use of other kind
of channels, like misdn as matteo did..
Quote:
Also since I wrote codec_amr and format_amr, i gave chan_zap the amr
capability so the call is transcoded.
Writting a codec_amr and format_amr would allow every channel to be able to
transcode them.
By the way, do you have them avaiable? Where can i donwload them? How can i
use your library with chan zap?
Quote:
Currently when I bridge to SIP it works but there is a BIG problem, it's
the FPS that the softphone is sending.
Quote:
Basically, if the H263 video coded by the softphone is about 50kb/s its
fine. but if it's bigger (which is usually the case) then you loose frame.
Quote:
Also, the softphone is not usually capable of sending a SIP info as Video
Fast Update asking the softphone to reencode an I-frame.
Quote:
That's why I'm currenty working on the transcoding of H263 <> H263.
Yes, you're rigth, most videophones doesn't allow you to adjust the bandwith
it's send, and if they send you more it wont work.
Transcoding can be done in to ways, the good one and the easy one. The easy
one is just use an h263 decoder and a h263 encoder, the good one,
as somone pointed in the past, would be reuse the motion vectors and some
other data from the decoder so the encoder doesn't need to calculate them
again. Which one are u implementing? And once again, where can i download
it?
To: asterisk-video@lists.digium.com
Subject: Re: [Asterisk-video] H324M AMR audio working!!!
Date: Thu, 1 Mar 2007 16:58:56 +0100
Hi Ratmin, it's good to see you're alive and well...
> From: Ramtin Amin
> Sent: Thursday, March 01, 2007 3:48 PM
>
> The problem with bridging to SIP is bigger than you think
>
> My H324m stack is directly implemented into chan_zap, so I can place 3G
call with dial(zap/gv1/number) or I receive video call from a cellphone in
H263 mode.
Good, and what it's the difference bridging the call to a sip channel?
Beside of that sepparating the stack from zap allows the use of other kind
of channels, like misdn as matteo did..
> Also since I wrote codec_amr and format_amr, i gave chan_zap the amr
capability so the call is transcoded.
Writting a codec_amr and format_amr would allow every channel to be able to
transcode them.
By the way, do you have them avaiable? Where can i donwload them? How can i
use your library with chan zap?
> Currently when I bridge to SIP it works but there is a BIG problem, it's
the FPS that the softphone is sending.
> Basically, if the H263 video coded by the softphone is about 50kb/s its
fine. but if it's bigger (which is usually the case) then you loose frame.
> Also, the softphone is not usually capable of sending a SIP info as Video
Fast Update asking the softphone to reencode an I-frame.
> That's why I'm currenty working on the transcoding of H263 <> H263.
Yes, you're rigth, most videophones doesn't allow you to adjust the bandwith
it's send, and if they send you more it wont work.
Transcoding can be done in to ways, the good one and the easy one. The easy
one is just use an h263 decoder and a h263 encoder, the good one,
as somone pointed in the past, would be reuse the motion vectors and some
other data from the decoder so the encoder doesn't need to calculate them
again. Which one are u implementing? And once again, where can i download
it?
Posted: Fri Mar 02, 2007 8:26 am Post subject: [Asterisk-video] H324M AMR audio working!!!
Hi,
On Thu, 2007-03-01 at 14:48 +0000, Ramtin Amin wrote:
Quote:
The problem with bridging to SIP is bigger than you think
My H324m stack is directly implemented into chan_zap, so I can place
3G call with dial(zap/gv1/number) or I receive video call from a
cellphone in H263 mode.
uhm I prefer Sergio's way... I don't like to be tied
to a single channel driver (and also zap is already a mess
of protocols...) but is bettere imho having a generic
stream processor that don't looks at the transport.
Quote:
Also since I wrote codec_amr and format_amr, i gave chan_zap the amr
capability so the call is transcoded.
That's nice. Can we try it ?
Quote:
Currently when I bridge to SIP it works but there is a BIG problem,
it's the FPS that the softphone is sending.
Basically, if the H263 video coded by the softphone is about 50kb/s
its fine. but if it's bigger (which is usually the case) then you
loose frame.
Also, the softphone is not usually capable of sending a SIP info as
Video Fast Update asking the softphone to reencode an I-frame.
That's why I'm currenty working on the transcoding of H263 <> H263.
I agree.
Keep us updated on this
regards,
Matteo.
--
Matteo Brancaleoni
R&D Director
Tel :+39.02.70633354
Voip :sip:matteo@sip.voismart.it
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