[Asterisk-Users] OH323 Channel Driver buffer sizes
Hemant Kumar
hkumar at spgsolutions.com
Fri May 9 10:56:54 MST 2003
whats the audiomode and connectmode you are using ?
----- Original Message -----
From: "Michael Manousos" <manousos at inaccessnetworks.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, May 09, 2003 10:48 PM
Subject: Re: [Asterisk-Users] OH323 Channel Driver buffer sizes
> Niclas Gustafsson wrote:
> > Hello!
> >
> > Anyone with some insight into the oh323 channel driver please shed
> > some light on the code block below from wrapendpoint.cxx.
> >
> > When enabling trace on the channel driver i get this, for me, strange
> > debug info:
> >
> > WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 320
> > WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits
> > 8
> > WrapH323EndPoint::OpenAudioChannel: Frame 1
> > WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k
> > WrapH323EndPoint::OpenAudioChannel: The sound channel is
> > audiosocket:out1(fd=41)
> > WrapH323EndPoint::OpenAudioChannel: The audio device name is
> > audiosocket:out1
> > PAsteriskSoundChannel::Open: os_handle 41, mediaFormat 0, frameTime 1
> > WrapH323EndPoint::OpenAudioChannel: Opened sound channel
> > "audiosocket:out1" for playing using 1x8 byte buffers.
> > WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [27258]
> > : receiving G.711-uLaw-64k{hw}
> > WrapH323EndPoint::OnStartLogicalChannel: RxFrames = 20
> > WrapH323EndPoint::OnStartLogicalChannel: channelsOpen = 1
> > WrapH323EndPoint::OpenAudioChannel: Direction => RECODER, Buffer => 320
> > WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits
> > 8
> > WrapH323EndPoint::OpenAudioChannel: Frames 20
> > WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k
> > WrapH323EndPoint::OpenAudioChannel: The sound channel is
> > audiosocket:in1(fd=43)
> > WrapH323EndPoint::OpenAudioChannel: The audio device name is
> > audiosocket:in1
> > PAsteriskSoundChannel::Open: os_handle 43, mediaFormat 0, frameTime 1
> > WrapH323EndPoint::OpenAudioChannel: Opened sound channel
> > "audiosocket:in1" for recording using 1x160 byte buffers.
> >
> > How come the buffers are of different size? Is this a good thing?
>
> These buffers are used to copy bytes between Asterisk and OpenH323
> lib. The two streams are completely unrelated. The logic behind
> this, is that the driver reads one frame each time (for sample-based
> codecs, a predefined number of samples is considered a frame),
> although there are (possibly) more data already received.
>
> > Recording
> > to 160 bytes and playing to 8 bytes buffers.... When looking into the
> > code,
> > it looks to me as if there has been some trial and error approach?
> >
> > (WrapH323EndPoint::OpenAudioChannel)
> >
> > /************
> >
> > bufSize = GetFrames(mediaFormat) *
> > mediaFormat.GetFrameSize();
> >
> > **********/
> > if (dir == RECORDER) {
> > bufSize = GetFrames(mediaFormat) *
> > mediaFormat.GetFrameSize();
> > WRAPTRACE(5, "Frames " <<
> > GetFrames(mediaFormat));
> > } else {
> > /*************
> >
> > if ((mediaType == RTP_DataFrame::PCMU) ||
> > (mediaType == RTP_DataFrame::PCMA)) {
> >
> > bufSize = bufferSize;
> >
> > } else {
> >
> > ***********/
> > bufSize = mediaFormat.GetFrameSize();
> > WRAPTRACE(5, "Frame 1");
> > /************
> >
> > }
> >
> > ***********/
> >
> > Shouldn't the buffers be of equal size?
>
> Not necessarily.
>
> > I'm quite new to this code so please correct me if i'm wrong,
> >
> > Ah, and yes, the reason i'm digging in the code is to get my fax to
> > work! ;) It worked fine
> > with the config CiscoGW -> GnuGK -> ATA186 -> Fax, but now it does not
> > and I get comm error
> > on the display. My config now looks like CiscoGW -> GnuGK -> Asterisk ->
> > ATA186 -> Fax. Any
> > ideas anyone, i've disable the silence suppression and disabled the CED
> > tone detection as
> > per Cisco "Using FAX Mode" in their Administrators Guide for the ATA186.
>
> You need T.38 support to be able to handle fax and, at this time,
> asterisk-oh323 doesn't support it. I 'll try to add some preliminary
> code in order to be able to test it.
>
> >
> > Regards,
> > Niclas Gustafsson
> >
> >
>
> Michael.
>
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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