On 06/03/2018 06:29, Phil Karn wrote:
How hard would it be for WJST to accept receive audio from a RTP (Real
Time Protocol) multicast network stream?
Hi Phil,
not very hard at all. The audio in WSJT-X is via a Qt I/O Device
abstraction and making a UDP server that joins a multicast group to
receive and send audio would only require a thin layer to manage
datagram reception and transmission converting into/out of a stream
in/out of respective QIODevice sub-classes is about the whole
requirement. The Modulator and Detector classes in WSJT-X are pretty
much agnostic about which QIODevice derivative they talk to for PCM
sample streams.
Can you point me to somewhere I can get information about gathering RTP
datagrams and combining as a continuous stream? Do I need to work with
RTCP as well?
Our experience of users using the Elecraft remote kit using IP streaming
is that latency and delay are a problem, this being because of our
dependence on external wall clock synchronization. Can RTP provide
absolute time-stamp information that we can leverage to capture UTC time
sync relative to the source? Is there a recovery mechanism to "unbuffer"
when delays accumulate?
I assume the RTP payload formats are usually compressed, can we use
uncompressed PCM at 48000Hz 16-bit (actually 12000Hz 16-bit is all we
require unless we go to wider bandwidths) and still expect timely
delivery? If not, are we heading for a world of pain with proprietary
codecs?
I have some experience as I used to work on a set top box application
that streamed MPEG-4 over ADSL Internet, we used the Intel IPP libraries
for codecs rendering to a Linux frame buffer and sending either
web-camera or Ethernet based stand-alone camera streams.
73
Bill
G4WJS.
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