Hi all,
We try to build a VoIP application for simulating several audio media
features on top of the pjsip stack. This application should have the
following features related to the pjsip-stack:
- support multiple ports of a jack client
- connection between transport streams and ports of a jack client (no
mixing)
In my opinion there are two different approaches to a solution, every one
leads to several poblems I want to describe in relation to the model used.
I guess, that a solution would require significant changes/extensions
within the pjsip stack.
*** Recording model (suitable in both models): ***
Jack client x input port - conferenc bridge master port - tx to all
transport streams (that's already the case)
*** Playback models: ***
Model 1
======================================================================
- use 1 jack input client and connect it to the conferenc bridge master
port
- use n-1 jack clients with 1 output port each and register them to
different ports at the conference bridge.
- That would require a support of n sounddevices simultaneously (todo).
Jack client 1 output port - conferenc bridge - rx transport stream 1
Jack client 2 output port - conferenc bridge - rx transport stream 2
Jack client 3 output port - conferenc bridge - rx transport stream 3
...
Jack client n-1 output port - conferenc bridge - rx transport stream n-1
Issues:
- As I see there is currently only one sounddevice supported by the
conderence bridge. This sounddevice has been selected within collected
device information gathered from hostapi/portaudio.
- Setting a new device will clear !! the previous one and register the new
one to the master conference port.
Solution:
- change the function pjsua_set_snd_dev() and related functions to handle
more than one sounddevice at once
- mirror the master port creation to other conference ports. (I don't know
if this is possible using the same get_frame/put_frame callbacks of the
master port.)
- assign every sounddevice a separate conference port
- Ensure that sounddevce functions like OpenStream() can be called on
different sounddevices simultaneously
Model 2
======================================================================
- use a jack client with one input port and n-1 output ports and connect
it to the conference bridge master port.
- That would require support of multiple channels from PortAudio to
conference bridge (todo)
Jack client x output port 1 - conferenc bridge - rx transport stream 1
Jack client x output port 2 - conferenc bridge - rx transport stream 2
Jack client x output port 3 - conferenc bridge - rx transport stream 3
...
Jack client x output port n-1 - conferenc bridge - rx transport stream n-1
Issues:
Starting from Jack I retraced the output port/channel information to the
conference bridge. I figured out that the information about multiple
ports/channels of a sounddevice gets lost in the transition from portaudio
(functions NonAdaptingProcess/AdaptingOutputProcess) to internal
functions like PaPlayerCallback. It looks like only the outputbuffers of the
first port/channel will be processed within pjspip . (variable void *output,
array which holds per channel output-buffers on the PortAudio-side)
Solution:
- provide, handle the channel / outputbuffer information from PortAudio to
internal processing structures
- disable mixing in conference bridge
- connect output ports with transport streams (function get_buffer())
(I have no idea how I can resolve this using the current structures).
Any thoughts about that, did anybody something similar before?
--
Thanks And Regards,
Thomas Kluge
