Dear all,
I use a Linux platform and my aim was to modify simple_pjsua program to
check if I have understood pj_port behaviour, but my modified program
doesn't work as I expected, so I ask here if anyone could help me in
understanding my mistakes, also thinking that this could help other
peoples to understand pj_port behaviour.
As you can see by reading my code at the end of this email, I have
substituted audio devices with "null sound device" (masterport with a
null upstream and downstream connected to port0 of conference bridge),
thinking it will act as a data-pump for added ports connected to port 0
of conference bridge, by delivering a proper clock to force data flow.
Then I have created my bidirectional ussd_port, with callbacks needed to
provide frame flow. Both callbacks have an instruction to log a single
sample from frame buffer (thinking it should change from a sound frame
to another), frame buffer size and an index (P, G) to track which
callback is currently running.
I have finally added my port to a free slot of conference bridge. Than I
have connected this slot to port 0 in both direction.
As you can see I have also bridged (though a global buffer copied by
callbacks) audio data flow coming from remote user agent to send
received frames back to remote user agent, thinking I would be able to
hear the echo of my voice in remote user agent.
When I start this program, it seems to correctly establish a sip
communication with remote user-agent (Linphone rings and shows
communication is on). Callbacks logs shows that my ussd_port is
correctly clocked, since there is a trace of callbacks activity, as I
expected.
....
09:30:08.374 myapp P size:640 b:0
09:30:08.374 myapp G size:640 b:0
09:30:08.393 myapp P size:640 b:0
09:30:08.394 myapp G size:640 b:0
....
But received audio data frames are filled only with zeros, even when I
speak in remote user-agent microphone. I can see microphone audio bar
movement in Linphone, but I don't see any movements in Linphone's
speaker bar and I don't hear any sound either, in remote user agent!
I have also tried to force a noise in remote user-agent by modified
outgoing frames sent by put_frame callback, but still I heard nothing in
remote user-agent speaker, nor its speaker sound bar seems to move.
To see if communication was correctly established, I made an attempt by
adding dmft tone capture callback in my program, and I can see that my
program correctly capture key pressed in remote user-agent
What is wrong in my code? Why audio frames are always zeroed?
For shortness, I haven't reported all simple_pjsua.c program code, but
just a bit of it to show where I have inserted my code I have also
reported port creation function code and get/put callback code. To write
my code I have been inspired by sinewave.c example. From sinewave.c I
have just copied but not fully understood lines marked with "///".
Thank you in advance to anyone that will give me any suggestion!
Best regards from Italy!
Silvio
------------------------------
PS: Here is my C code:
------------------------------
int main(int argc, char *argv[])
{
.......
.......
/* Init pjsua */
{
pjsua_config cfg;
pjsua_logging_config log_cfg;
pjsua_config_default(&cfg);
cfg.cb.on_incoming_call = &on_incoming_call;
cfg.cb.on_call_media_state = &on_call_media_state;
cfg.cb.on_call_state = &on_call_state;
pjsua_logging_config_default(&log_cfg);
log_cfg.console_level = 5;
status = pjsua_init(&cfg, &log_cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
}
/* I have added this code */
{
PJ_LOG(3,(THIS_FILE, "conf bridge active port number %d",
pjsua_conf_get_active_ports()));
// Create master port, connecting port0 of the conference bridge to
a null port
// null port is upstream port, conference bridge port0 is connected
to downstream port
// Null sound device will act as a frame streaming engine for other
ports connected to
// conference bridge, by clocking associated callbacks
pjsua_set_null_snd_dev();
PJ_LOG(3,(THIS_FILE, "null snd device set, conf bridge active port
number %d", pjsua_conf_get_active_ports()));
// Create pool for application
pj_pool_t *app_pool;
app_pool = pjsua_pool_create("app_pool", 1000, 1000); // TODO
capire cosa bisogna mettere
// Create bidirectional media port
pjmedia_port *ussd_port;
status = create_ussd_port(app_pool, &ussd_port);
if (status != PJ_SUCCESS) error_exit("Error unable to create ussd
port", status);
PJ_LOG(3,(THIS_FILE, "ussd, bidirectional user space sound device
port, created"));
// ask for a free slots in conference bridge
pjsua_conf_port_id ussd_conf_slot_id;
// add ussd port to conference bridge
// pj_status_t pjsua_conf_add_port (pj_pool_t *pool, pjmedia_port
*port, pjsua_conf_port_id *p_id)
status = pjsua_conf_add_port(app_pool, ussd_port, &ussd_conf_slot_id);
if (status != PJ_SUCCESS) error_exit("Error unable to add ussd port
to conference bridge", status);
// from now the conference port slot connected to ussd will be
identified by ussd_conf_slot_id
PJ_LOG(3,(THIS_FILE, "ussd port added to conf bridge, conf bridge
active port number %d", pjsua_conf_get_active_ports()));
// ussd port has been created and added to a slot of conf bridge,
but there isn't any data flow yet
// since it is not connected: so now we need to connect ussd conf
slot to port 0 of conference bridge,
// providing a connection for each data direction
status = pjsua_conf_connect(0, ussd_conf_slot_id);
if (status != PJ_SUCCESS) error_exit("Error unable to connect ->
ussd port to conference bridge", status);
status = pjsua_conf_connect(ussd_conf_slot_id,0);
if (status != PJ_SUCCESS) error_exit("Error unable to connect <-
ussd port to conference bridge", status);
PJ_LOG(3,(THIS_FILE, "ussd port connected to conf bridge:
bidirectional ussd data flow is up"));
}
/* end of my added code */
/* Add UDP transport */
{
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
cfg.port = MY_UDP_PORT;
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error creating transport",
status);
}
....
....
}
/* I have added a global buffer and costants */
pj_int16_t g_buf[AUDIO_SPF+2]; // just a little bigger than needed
// audio bits per sample
#define AUDIO_BPS 16
// audio channels number
#define AUDIO_CHN 1
// audio samples per second
#define AUDIO_SPS 16000
// audio sample per frame
#define AUDIO_SPF 320
/* I have also added these functions: */
/* callback */
static pj_status_t ussd_get_frame (pjmedia_port *port, pjmedia_frame *frame)
{
/* a sound frame requires is trasferred from g_buf to port */
pj_int16_t *samples = frame->buf;
pj_memcpy(frame->buf, g_microphone_buf, frame->size);
frame->type = PJMEDIA_FRAME_TYPE_AUDIO;
PJ_LOG(3,(THIS_FILE, "P size:%d b:%d",frame->size,samples[10])); //
test
return PJ_SUCCESS;
}
/* callback */
static pj_status_t ussd_put_frame (pjmedia_port *port, pjmedia_frame *frame)
{
/* a sound frame requires is trasferred from port to g_buf */
pj_int16_t *samples = frame->buf;
pj_memcpy(g_buf, frame->buf, frame->size);
PJ_LOG(3,(THIS_FILE, "G size:%d b:%d",frame->size,samples[10])); //
test
return PJ_SUCCESS;
}
/* Create ussd bidirectional port */
static pj_status_t create_ussd_port(pj_pool_t *pool, pjmedia_port **p_port)
{
pjmedia_port *port;
unsigned i;
unsigned count;
unsigned channel_count=AUDIO_CHN;
unsigned bits_per_sample=AUDIO_BPS;
unsigned sample_per_frame=AUDIO_SPF;
unsigned sampling_rate=AUDIO_SPS;
pj_str_t name;
PJ_ASSERT_RETURN(pool && channel_count > 0 && channel_count <= 2,
PJ_EINVAL); ///
port = pj_pool_zalloc(pool, sizeof(pjmedia_port));
PJ_ASSERT_RETURN(port != NULL, PJ_ENOMEM); ///
// Fill in port info.
name = pj_str("ussd");
pjmedia_port_info_init( &port->info, // (pjmedia_port_info *)
&name,
PJMEDIA_SIG_CLASS_PORT_AUD('s', 'i'), ///
(unsigned) signature
sampling_rate, // (unsigned) clock rate
channel_count, // (unsigned) channel count
bits_per_sample, // (unsigned) bits_per_sample,
sample_per_frame); // (unsigned)
samples_per_frame
// Set the functions to feed ussd frame in both direction
port->get_frame = &ussd_get_frame;
port->put_frame = &ussd_put_frame;
*p_port = port;
return PJ_SUCCESS;
}
