Hello,
I am working on a project on an embedded linux device running RHEL7 where two calls are established with PJSIP, each with a seperate mono audio stream at 8kHz. We want one call to be routed to the left channel of our audio device and the other call to go to the right channel. To accomplish this, we created a splitter/combiner as per the stereo example in the pjsua program. However, when setting up audio this way, we see a lot of problems. We see very high jitter values on the TX side of the calls, 1500-2000 msec. Perhaps because of this, the incoming audio is very distorted in both the left and right channel. The left channel sounds low and garbled, and has a latecy of around 1 second. The right channel is also garbled, but can be understood and sounds like it is at a higher pitch. Below is a portion of the logs showing the audio issue:
09:22:00.612 scombdb-up !Underflow, buf_cnt=0, will generate 1 frame
09:22:00.682 scombdb-dn !304 samples reduced, buf_cnt=1206
09:22:00.682 scombdb-dn Buffer size adjusted from 1510 to 1206 (eff_cnt=960)
09:22:00.697 scombdb-up !Underflow, buf_cnt=0, will generate 1 frame
09:22:00.767 scombdb-dn !336 samples reduced, buf_cnt=1190
09:22:00.767 scombdb-dn Buffer size adjusted from 1526 to 1190 (eff_cnt=960)
09:22:00.782 scombdb-up !Underflow, buf_cnt=0, will generate 1 frame
09:22:00.847 scombdb-up Underflow, buf_cnt=0, will generate 1 frame
09:22:00.851 scomb-rev !Pausing media flow on downstream direction (level=14)
09:22:00.888 scomb-rev !Resuming media flow on downstream direction (level=13)
09:22:00.932 scomb-rev Pausing media flow on upstream direction (level=-14)
09:22:00.975 scomb-rev Resuming media flow on upstream direction (level=-13)
09:22:01.001 scombdb-dn !403 samples reduced, buf_cnt=1427
09:22:01.002 scombdb-dn Buffer size adjusted from 1830 to 1427 (eff_cnt=960)
09:22:01.017 scombdb-up !Underflow, buf_cnt=0, will generate 1 frame
09:22:01.087 scombdb-dn !235 samples reduced, buf_cnt=1512
Below is the code used to connect the audio device to the conference bridge via the splitcomb:
#define LEFT_CH 0
#define RIGH_CH 1
#define NO_OPT 0
// Set up any variables
pjmedia_port *conf;
pj_pool_t* splitterPool;
splitterPool = pjsua_pool_create("splitterPoo
pjmedia_port *splitcomb;
pjmedia_port *revChPort;
pjsua_conf_port_id rightPortId;
pjmedia_snd_port *sndDevPort;
/* Disable existing sound device and get conf master port.*/
conf = pjsua_set_no_snd_dev();
/* Create stereo-mono splitter/combiner */
status = pjmedia_splitcomb_create(split
pj_assert(status == PJ_SUCCESS);
/* Connect channel0 (left channel?) to conference port slot0 */
status = pjmedia_splitcomb_set_channel(
pj_assert(status == PJ_SUCCESS);
/* Create reverse channel for channel1 (right channel?)... */
status = pjmedia_splitcomb_create_rev_c
pj_assert(status == PJ_SUCCESS);
/* .. and register it to conference bridge (it would be slot1
* if there's no other devices connected to the bridge)
*/
status = pjsua_conf_add_port(splitterPo
pj_assert(status == PJ_SUCCESS);
/* Create sound device */
status = pjmedia_snd_port_create(splitt
pj_assert(status == PJ_SUCCESS);
/* Connect the splitter to the sound device */
status = pjmedia_snd_port_connect(sndDe
pj_assert(status == PJ_SUCCESS);
When calls are made we then call pjsua_conf_connect(callId1, LEFT_CH) and pjsua_conf_connect(callId2, RIGHT_CH) for the respective calls.
Is there any step missing in the above code to set up the splitter/combiner? Do the audio logs shown above indicate a specific problem?
What I've done so far: I have tried adjusting the values used for pjmedia_splitcomb_create(...) and pjmedia_snd_port_create(...) manually, but audio quality did not improve. I also disbaled AEC with no improvement. The processor sits at 90% idle according to top.
Thanks,
John
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