Changes between Initial Version and Version 1 of Ticket #323


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Timestamp:
Jun 6, 2007 3:01:59 PM (17 years ago)
Author:
bennylp
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  • Ticket #323 – Description

    initial v1  
    11!ChenHuan sent this code to the list on 2007/05/06, along with this mail: 
    22{{{ 
    3 Currently, it only uses PCMA & PCMU codec, and the quality of the voice on Windows Mobile is almost as good as on PC's 
     3Currently, it only uses PCMA & PCMU codec, and the quality of the  
     4voice on Windows Mobile is almost as good as on PC's 
    45  
    56These are the key points I conclude: 
    67  
    7 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. 
     81. I did not use the speaker thread to stimulate the sending of RTP  
     9packets, but used another clock thread. There are many methods to  
     10achieve this, but my method is to change the file : conference.c.  
     11The advantage is that we have not to modify other files including  
     12the program on PJSIP. 
    813  
    9 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. 
     14I added a pjmedia_clock and a speaker thread buffer member into  
     15struct conf_port. Particularly, the speaker buffer is similar to the  
     16mic thread buffer. The function get_frame just only reads frames  
     17from the speaker buffer, and leaves other things to the  
     18clock_callback function. 
    1019  
    1120The attachment is my conference.c 
    1221  
    13222. Don't use any ACE 
    14 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. 
     23Simple ACE is not recommended, because it will cause the outgoing  
     24voices quite instable, which means the volume of the sound  
     25alternates between low and high. Whether you use the default  
     26PJMEDIA_ECHO_SUPPRESS_FACTOR 4 or 2, it did't work out. 
    1527  
    16 Also the normal ACE algorithm is too costly for my device, I have to give it up 
     28Also the normal ACE algorithm is too costly for my device, I have to  
     29give it up 
    1730  
    1831So, I suggest that ACE should be disabled on Windows Mobile 
    … …  
    2134#define PJMEDIA_ECHO_SUPPRESS_FACTOR        1 
    2235  
    23 After doing the two steps above, the quality of the voice will improved a lot, 
    24 and if you feel the quality is very bad when calling PSTN phones, maybe the following method will help 
     36After doing the two steps above, the quality of the voice will  
     37improved a lot, and if you feel the quality is very bad when calling  
     38PSTN phones, maybe the following method will help 
    2539  
    26403. Change the jitter buffer's parameters when necessary