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Niclas Gustafsson

2003-May-09 06:11 UTC

[Asterisk-Users] OH323 Channel Driver buffer sizes

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?
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?
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.

Regards,
Niclas Gustafsson

Michael Manousos

2003-May-09 10:18 UTC

head link

[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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

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