Jeremy A wrote:
Hi,
I've got a few questions and perhaps a problem or two with using codecs
in SEMS 1.2.1
Is there any rate conversion between files stored on disk in WAV format
and the line rate? In my case I have a 16000 Hz wav file and am using
G711 codec. The played out sound is very slow, so even though SEMS reads
the file as 16000 Hz (as per debug) The delivered rate assumes 8000 Hz
so the sound is half speed.
I am a bit surprised there is no rate conversion, so I would like to
sampling rate conversion is currently only done in the wb (widenband)
branch, which supports different sample rates.
know if there is a flag I have to set?
I am trying to use G726-32 codec for some communications. The behaviour
is very odd
The debug shows the call phone requesting G726_32
Content-Type: application/sdp
P-App-Param: usr=12345
v=0
o=- 7373316 7373316 IN IP4 203.31.40.160
s=-
c=IN IP4 203.31.40.102
t=0 0
m=audio 8020 RTP/AVP 2 101
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
(I'm using rtpproxy for recording which causes a minor and inaccurate
problem from the nortpproxy line)
The reply is
Content-Type: application/sdp
Content-Length: 198
v=0
o=sems 1219929049 1248189210 IN IP4 203.31.40.68
s=session
c=IN IP4 203.31.40.68
t=0 0
m=audio 8000 RTP/AVP 2 101
a=rtpmap:2 G721/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Where it seems SEMS has substituted G721 for G726_32 ? Is this an
error? (This shows up on wireshark as well)
it's the same codec. g726 is the newer standard. for an overview, see
e.g. wikipedia.
At this stage I get a protocol error on the phone.
that's not good. which one is it? IMO the phone should not complain.
quick fix: swap lines 94 and 95 of core/plug-in/adpcm/adpcm.c, or
remove line 94 (but then sems won't understand G726-32)
At various stages of testing I've got sound through but just as noise.
Does SEMS require the wav file to be in the same format as the line
format? or can it transcode? Can it rate convert when codecs are different.
it does transcoding. the best is to have wav files as 16 bit pcm, 8khz
sample rate. otherwise, sems needs to transcode into 16 bit pcm, and
then into the codec used in the call.
I can work around all these problems by using G711 and 8000 Hz wav
files, but being able to work at G726-32 would be very helpful, as would
don't use G711 for the wav files, but linear PCM 16 bit. otherwise you
loose audio quality (G711 is NOT transparent, and you'd effectively
tandem G711 and G726).
obcious exception: precoded announcements.
automatic rate conversion.
indeed, that would be convenient. but already for performance reasons
alone, I would resample the files offline.
hth
Stefan
Thanks in advance
Jeremy
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--
Stefan Sayer
VoIP Services Consulting and Development
Warschauer Str. 24
10243 Berlin
tel:+491621366449
sip:[email protected]
email/xmpp:[email protected]
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