Code to use different clock for the bridge (thanks ChenHuan)

[bennylp]

ChenHuan sent this code to the list on 2007/05/06, along with this mail:

Currently, it only uses PCMA & PCMU codec, and the quality of the voice on Windows Mobile is almost as good as on PC's
 
These are the key points I conclude:
 
1. I did not use the speaker thread to stimulate the sending of RTP packets, but used another clock thread. There are many methods to achieve this, but my method is to change the file : conference.c. The advantage is that we have not to modify other files including the program on PJSIP.
 
I added a pjmedia_clock and a speaker thread buffer member into struct conf_port. Particularly, the speaker buffer is similar to the mic thread buffer. The function get_frame just only reads frames from the speaker buffer, and leaves other things to the clock_callback function.
 
The attachment is my conference.c
 
2. Don't use any ACE
Simple ACE is not recommended, because it will cause the outgoing voices quite instable, which means the volume of the sound alternates between low and high. Whether you use the default PJMEDIA_ECHO_SUPPRESS_FACTOR 4 or 2, it did't work out.
 
Also the normal ACE algorithm is too costly for my device, I have to give it up
 
So, I suggest that ACE should be disabled on Windows Mobile
You can do it by this way:
#define PJMEDIA_HAS_SPEEX_AEC               0
#define PJMEDIA_ECHO_SUPPRESS_FACTOR        1
 
After doing the two steps above, the quality of the voice will improved a lot,
and if you feel the quality is very bad when calling PSTN phones, maybe the following method will help
 
3. Change the jitter buffer's parameters when necessary
Supposing you are using pjsua, you can do it by this:
    app_config.media_cfg.jb_max     = 80;
    app_config.media_cfg.jb_init    = 40;
    app_config.media_cfg.jb_min_pre = 20;
    app_config.media_cfg.jb_max_pre = 80;
These codes can be put on the tail of func : default_config
 
 
I hope it will help.

ChenHuan
2007-06-06 20:01:10
chenhuan@sict.ac.cn