Hi James,
Im asking you becuae looking thru the archives you asked a similar question a long time ago.
i'm using an open source VoIP application with the
alsa driver. My card is the onboard intel8x0.
i'm using an open source VoIP application with the
alsa driver. My card is the onboard intel8x0.
My problem is figuring out the patterns I am
getting with the alsa driver when transmitting
packets.
For instance when I use the GSM codec with a packets
size of 80 msecs and 132 bytes, packets are
transmitted at alternate intervals of 64/96 msecs.
When i use G.711 packets of 240 bytes are transmitted
getting with the alsa driver when transmitting
packets.
For instance when I use the GSM codec with a packets
size of 80 msecs and 132 bytes, packets are
transmitted at alternate intervals of 64/96 msecs.
When i use G.711 packets of 240 bytes are transmitted
every 32 msecs with an extra packet every 15 pkts.
With G.711 it is obvious that 256 bytes are added to driver sending buffer every 32 msecs.However this does not match for GSM because it needs only 132 bytes every 64/96 msecs.
Can the driver change the amount of data it gets from the soundcard to suit the codec?
Help appreciated,
Brian.
James Courtier-Dutton <[EMAIL PROTECTED]> wrote:
Brian.
James Courtier-Dutton <[EMAIL PROTECTED]> wrote:
Brian Furey wrote:
> Hi all,
> I have an intel810 onboard soundcard.I am using the
> alsa driver with a VoIP session.
> The intel8x0.c file has a minimum period byte size
> of 32 bytes with the minimum no. of periods being
> 1.The min and max rate is set to 48k.
>
> How can I find out what actual(runtime) size period
> the alsa driver is dealing with?
>
> Does it use the minimum size as the period size?
>
> Brian.
>
Brian, I am working on a VoIP setup. I am updating the asterisk alsa
console driver so that it actually works! The current driver is stuck
round about alsa api 0.5.x
The period size that the sound card is actually using does not really
matter, if just effects latency. The bigger the period size, the higher
the latency.
Just set the period size to t he smallest the sound card can do, and then
just read and write to it.
I found that PLAYBACK and CAPTURE directions can have different period
sizes, so it is better to open separate handles for playback and capture.
In the config setup, you set the buffer and period sizes, and before you
set them, you can retrieve the current min/max period and buffer sizes.
For playback, it is best to have a certain minimum buffer being full
most of the time, due to network jitter, and this buffer can act as the
jitter buffer. The actual size of this is probably best found out from
trial and error (I have not finished testing this bit yet).
For capture, just poll for input, and then read whatever is in the
capture buffer and transmit it. You can experiment with different
methods of early buffer reads, but again, I have not finished testing
with that, so I can't give you any definite answers.
Another thing to test with could be sampling and playback at 48k.
Most sound cards work at 48k, and this would reduce the period_time.
(period_size stays the same, but as one is using a higher rate, the
period_time decreases, and thus latency.) The problems with that is that
most VoIP is at 8k, so some resampling is required.
Cheers
James
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