Hi Joao,
Please see inline comments..
2008/8/22 Jo?o C?sar <jpcesar at gmail.com>:
> Hi Nanang,
>
> these are the last values before the assertion fails:
>
> 256 <= 320
> 160 <= 160
> 352 <= 320
Those values (256 & 352) seems odd, what is the value of
'memory_port'->info.samples_per_frame? And also conference bridge's
samples_per_frame value? Could you send along the result of 'cl'
command in pjsua prompt?
>
> Assertion failed: cport->rx_buf_count <= cport->rx_buf_cap, file
> e:\development\
> jcesar\pjproject-0.9.0\pjmedia\src\pjmedia\conference.c, line 1404
>
>> Which codec and which clock rates are you using for
>> the sound device/conference and conference ports used such as file
>>ports if any. And, sometime reclean the whole project may resolve a
>> problem occured after changing library.
>
> we are working with the no sound device option since we only use PJSIP to
> deliver the packets to a third party application via SIP. we receive audio
> packets from a socket, if we dont have a call for that client we make the
> call, else we just feed the memory port with more samples.
>
> we are making the call like this:
>
> sampleRate = 8000;
> channels = 1;
> number_frames = 960;
> channel_count = 1;
>
> // create a media port to generate samples.
> status = create_memory_port( pool, sampleRate, 512, number_frames,
> channel_count, &callData[sessionId].memory_port_phone_out);
>
What is 512 stand for here?
> // add port to conference
> status = pjsua_conf_add_port(pool,
> callData[sessionId].memory_port_phone_out, &callData[sessionId].port_id);
> if (status != PJ_SUCCESS) error_exit("Error adding memory port", status);
>
> status = pjsua_call_make_call(acc_id,uri, 0, &callData[sessionId].port_id,
> NULL,NULL);
> if (status != PJ_SUCCESS) error_exit("Error making call", status);
>
> // assign the callid to the current call in callData
> callData[sessionId].call_id = callId;
>
> // increment callId for next call
> callId++;
>
> //creates the sound port
> status = pjmedia_snd_port_create_player(pjsua_var.pool,
> 0,
>
> 8000,
>
> pjsua_var.mconf_cfg.channel_count,
>
> pjsua_var.mconf_cfg.samples_per_frame,
>
> pjsua_var.mconf_cfg.bits_per_sample,
> 0,
> &sound_port);
>
> status = pjmedia_snd_port_connect(sound_port,
> pjmedia_conf_get_master_port(pjsua_var.mconf));
>
>
> Each time we receive an audio packet we put that frame into memory_port:
>
> // put frame
> memory_put_frame(callData[sessionId].memory_port_phone_out, &pjframe );
>
>
> and these are the options on the create memory port function:
>
>
> /* Fill in port info. */
> port->info.bits_per_sample = 16;
> port->info.channel_count = channel_count;
> port->info.encoding_name = pj_str("pcm");
> port->info.has_info = 1;
> port->info.name = pj_str("memory playback");
> port->info.need_info = 0;
> port->info.pt = 0xFF;
> port->info.clock_rate = sampling_rate;
> port->info.samples_per_frame = frame_size / 2 / channel_count ;
> port->info.bytes_per_frame = frame_size;
> port->info.type = PJMEDIA_TYPE_AUDIO;
>
Couldn't find any suspect from the snippet code since it shows mostly
variables without knowing the 'important' values. Actually I am
interested to know what the value of memory
port->info.samples_per_frame and conference's samples_per_frame.
>
> This was working fine on 0.8.
>
Conf port's buffer size (rx_buf_cap) calculation has been updated in
0.9 due to multichannel/stereo support in conference (and also IIRC it
seems rx_buf_cap in 0.8 is a bit over calculated, perhaps that's why
it was working fine on 0.8, miscalculation was tolerated).
Regards,
nanang
