[Asterisk-Dev] Any standard "recipe" for dealing with channel datarate mismatches?

[steve at daviesfam.org]

Hi,

I'm working on a bug with ever-increasing delay seen on a 3rd-party 
channel driver for Asterisk. (For the Sirrix basic-rate board).

This happens when a SIP channel is bridged to this channel driver.


The problem turns out to be that the SIP soft phone is sending audio
slightly faster than it leaves through the problem channel driver - about
2% too fast.

So the audio gradually accumulates in the buffers within the Sirrix
device's device driver.  And it had way too much buffering.

I've reduced that buffering.

But still, I have the problem that sooner or later, whatever size you use,
the write buffer to the device will get full.

Then what?

The options I can think of:

1) When the write fails because the buffer is full, just toss that frame 
only away.

The trouble with this is that the device driver buffer fills up and then 
stays nearly full.  If we are writing 20msec of audio at a time, we'll 
eventually settle in a pattern of regular and quite frequent glitches in 
the sent audio.  In this case, 20msec of audio is dropped every second or 
so, creating quite a nasty effect.

Plus, latency gets "stuck" at whatever buffer size the device driver is 
using.

or,

2) When the write fails because of full buffer, tell the device driver to 
flush the *entire* write buffer

We make one big, obvious audio drop every now and then - in this case 
every 26 seconds a "chunk" of 500msec of audio will be dropped.

or,

3) "Resample" the audio

I guess this is the best approach - to use a resampling process to reclock 
the audio.  This will keep the two sides nicely in step with a 1 byte slip 
every 50 bytes in my example.

Is there such a resampler in the code anywhere?

What happens in other cases like this (eg SIP -> Zap)

Thanks and regards,
Steve Davies