struct my_call_data *call_init_tonegen(pjsua_call_id call_id)
{
pj_pool_t *pool;
struct my_call_data *cd;
pjsua_call_info ci;
if (call_id !=-1 ) {
pjsua_call_get_info(call_id, &ci);
if (ci.media_status != PJSUA_CALL_MEDIA_ACTIVE)
return NULL;
}
pool = pjsua_pool_create("mycall", 512, 512);
cd = PJ_POOL_ZALLOC_T(pool, struct my_call_data);
cd->pool = pool;
pjmedia_tonegen_create(cd->pool, 8000, 1, 160, 16, 0, &cd->tonegen);
pjsua_conf_add_port(cd->pool, cd->tonegen, &cd->toneslot);
if (call_id !=-1 ) {
pjsua_conf_connect(cd->toneslot, ci.conf_slot);
}
if (accountSettings.localDTMF) {
pjsua_conf_connect(cd->toneslot, 0);
}
if (call_id !=-1 ) {
pjsua_call_set_user_data(call_id, (void*) cd);
}
return cd;
}
struct my_call_data *call_init_tonegen(pjsua_call_id call_id)
{
pj_pool_t *pool;
struct my_call_data *cd;
pjsua_call_info ci;
if (call_id !=-1 ) {
pjsua_call_get_info(call_id, &ci);
}
pool = pjsua_pool_create("mycall", 512, 512);
cd = PJ_POOL_ZALLOC_T(pool, struct my_call_data);
cd->pool = pool;
pjmedia_tonegen_create(cd->pool, 8000, 1, 160, 16, 0, &cd->tonegen);
pjsua_conf_add_port(cd->pool, cd->tonegen, &cd->toneslot);
if (call_id !=-1 ) {
pjsua_conf_connect(cd->toneslot, ci.conf_slot);
}
pjsua_conf_connect(cd->toneslot, 0);
if(call_id !=-1)
{
pjsua_call_set_user_data(call_id, (void*) cd);
}
return cd;
}
/* Callback called by PJSUA when call's media state has changed */
static void on_call_media_state(pjsua_call_id call_id) {
qDebug() << "Call media state";
pjsua_call_info ci;
if (pjsua_call_get_info(call_id, &ci) != PJ_SUCCESS) {
return;
}
pjsua_conf_connect(ci.conf_slot, 0);
pjsua_conf_connect(0, ci.conf_slot);
}
/* Callback called by the library when call's media state has changed */
static void on_call_media_state(pjsua_call_id call_id) {
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
// When media is active, connect call to sound device.
pjsua_conf_connect(ci.conf_slot, 0);
pjsua_conf_connect(0, ci.conf_slot);
}
}
/*
* Callback on media state changed event.
* The action may connect the call to sound device, to file, or
* to loop the call.
*/
static void on_call_media_state (pjsua_call_id call_id)
{
pjsua_call_info call_info;
pjsua_call_get_info (call_id, &call_info);
if (call_info.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
pjsua_conf_connect (call_info.conf_slot, 0);
pjsua_conf_connect (0, call_info.conf_slot);
}
}
void QPhoneHandler::onCallMediaState(qpjsua::CallInfo aCallInfo)
{
callInfoOut(aCallInfo);
if(aCallInfo.getMediaStatus() == PJSUA_CALL_MEDIA_ACTIVE) {
pjsua_conf_connect(0, aCallInfo.getConferenceSlot());
pjsua_conf_connect(aCallInfo.getConferenceSlot(), 0);
// view->setButtonText("Auflegen");
// connect(view, SIGNAL(accept()), SLOT(onHangup()));
}
}
//----------------------------------------------------------------------
void SipPhone::callMediaStateCb(pjsua_call_id call_id)
{
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE)
{
// When media is active, connect call to sound device.
pjsua_conf_connect(ci.conf_slot, 0);
pjsua_conf_connect(0, ci.conf_slot);
}
LogInfo info(LogInfo::STATUS_DEBUG, "pjsip", 0, "Call-media-state changed to "+QString::number(ci.state));
self_->signalLogData(info);
}
static void ui_conf_connect(char menuin[])
{
char tmp[10], src_port[10], dst_port[10];
pj_status_t status;
int cnt;
const char *src_title, *dst_title;
cnt = sscanf(menuin, "%s %s %s", tmp, src_port, dst_port);
if (cnt != 3) {
ui_conf_list();
src_title = (menuin[1]=='c'? "Connect src port #":
"Disconnect src port #");
dst_title = (menuin[1]=='c'? "To dst port #":"From dst port #");
if (!simple_input(src_title, src_port, sizeof(src_port)))
return;
if (!simple_input(dst_title, dst_port, sizeof(dst_port)))
return;
}
if (menuin[1]=='c') {
status = pjsua_conf_connect(my_atoi(src_port), my_atoi(dst_port));
} else {
status = pjsua_conf_disconnect(my_atoi(src_port), my_atoi(dst_port));
}
if (status == PJ_SUCCESS) {
puts("Success");
} else {
puts("ERROR!!");
}
}
// helper for creating call-media-player
static void create_player(pjsua_call_id call_id)
{
// get call infos
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
pj_str_t name;
pj_status_t status = PJ_ENOTFOUND;
log_message("Creating player ... ");
// create player for playback media
if (app_cfg.wav_file) {
status = pjsua_player_create(pj_cstr(&name, app_cfg.wav_file), 0, &play_id);
} else {
status = pjsua_player_create(pj_cstr(&name, app_cfg.tts_file), 0, &play_id);
}
if (status != PJ_SUCCESS) error_exit("Error playing sound-playback", status);
// connect active call to media player
pjsua_conf_connect(pjsua_player_get_conf_port(play_id), ci.conf_slot);
// get media port (play_port) from play_id
status = pjsua_player_get_port(play_id, &play_port);
if (status != PJ_SUCCESS) error_exit("Error getting sound player port", status);
// register media finished callback
status = pjmedia_wav_player_set_eof_cb(play_port, NULL, &on_media_finished);
if (status != PJ_SUCCESS) error_exit("Error adding sound-playback callback", status);
log_message("Done.\n");
}
//----------------------------------------------------------------------
bool SipPhone::addCallToConference(const int &call_src, const int &call_dest)
{
int src = call_src;
int dest = call_dest;
if (src == -1 || dest == -1)
{
LogInfo info(LogInfo::STATUS_ERROR, "pjsip", 0, "Error: one of the selected calls does NOT exist!");
signalLogData(info);
return false;
}
pjsua_call_info src_ci, dest_ci;
pjsua_call_get_info(src,&src_ci);
pjsua_call_get_info(dest,&dest_ci);
pj_status_t status = pjsua_conf_connect(src_ci.conf_slot,dest_ci.conf_slot);
if (status != PJ_SUCCESS)
{
LogInfo info(LogInfo::STATUS_ERROR, "pjsip", status, "Error connecting conference!");
signalLogData(info);
return false;
}
return true;
}
/*****************************************************************************
* test: play WAV file
*/
static void systest_play_wav(unsigned path_cnt, const char *paths[])
{
pjsua_player_id play_id = PJSUA_INVALID_ID;
enum gui_key key;
test_item_t *ti;
const char *title = "WAV File Playback Test";
pj_status_t status;
ti = systest_alloc_test_item(title);
if (!ti)
return;
pj_ansi_snprintf(textbuf, sizeof(textbuf),
"This test will play %s file to "
"the speaker. Please listen carefully for audio "
"impairments such as stutter. Let this test run "
"for a while to make sure that everything is okay."
" Press OK to start, CANCEL to skip",
paths[0]);
key = gui_msgbox(title, textbuf,
WITH_OKCANCEL);
if (key != KEY_OK) {
ti->skipped = PJ_TRUE;
return;
}
PJ_LOG(3,(THIS_FILE, "Running %s", title));
/* WAV port */
status = create_player(path_cnt, paths, &play_id);
if (status != PJ_SUCCESS)
goto on_return;
status = pjsua_conf_connect(pjsua_player_get_conf_port(play_id), 0);
if (status != PJ_SUCCESS)
goto on_return;
key = gui_msgbox(title,
"WAV file should be playing now in the "
"speaker. Press OK to stop. ", WITH_OK);
status = PJ_SUCCESS;
on_return:
if (play_id != -1)
pjsua_player_destroy(play_id);
if (status != PJ_SUCCESS) {
systest_perror("Sorry we've encountered error", status);
ti->success = PJ_FALSE;
pj_strerror(status, ti->reason, sizeof(ti->reason));
} else {
key = gui_msgbox(title, "Is the audio okay?", WITH_YESNO);
ti->success = (key == KEY_YES);
if (!ti->success)
pj_ansi_strcpy(ti->reason, USER_ERROR);
}
return;
}
void ringback_start(){
if (app_config.ringback_on) {
//Already ringing back
return;
}
app_config.ringback_on = PJ_TRUE;
if (++app_config.ringback_cnt==1 && app_config.ringback_slot!=PJSUA_INVALID_ID){
pjsua_conf_connect(app_config.ringback_slot, 0);
}
}
void ringback_start() {
if (css_var.ringback_on) {
//Already ringing back
return;
}
css_var.ringback_on = PJ_TRUE;
if (++css_var.ringback_cnt == 1 && css_var.ringback_slot != PJSUA_INVALID_ID) {
pjsua_conf_connect(css_var.ringback_slot, 0);
}
}
void SipAccount::onCallMediaState(pjsua_call_id call_id) {
Logger::debug("SipAccount::onCallMediaState(call_id=%d)...", call_id);
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
// connect active call to silence
pjsua_conf_connect(m_pPhone->getMediaConfSilenceId(), ci.conf_slot);
}
}
void QVDoorcom::onIncomingCall(int call_id) {
pj_status_t status;
if (active_call >= 0) {
return;
}
qDebug() << "incoming call" << call_id;
active_call = call_id;
w_dooropen->hide();
popup->show();
#ifdef DOORBELL_WAV
status = pjmedia_wav_player_port_set_pos(bell_file_port,0);
qDebug() << "Rewind status" << status;
if (status == PJ_SUCCESS) {
pjsua_conf_connect(bell_port_id,0);
// pjsua_conf_adjust_tx_level(0,1.0);
}
#else
pjsua_conf_connect(bell_port_id,0);
#endif
}
void MainWin::on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
for (unsigned i=0; i<ci.media_cnt; ++i) {
if (ci.media[i].type == PJMEDIA_TYPE_AUDIO) {
switch (ci.media[i].status) {
case PJSUA_CALL_MEDIA_ACTIVE:
pjsua_conf_connect(ci.media[i].stream.aud.conf_slot, 0);
pjsua_conf_connect(0, ci.media[i].stream.aud.conf_slot);
break;
default:
break;
}
} else if (ci.media[i].type == PJMEDIA_TYPE_VIDEO) {
emit signalInitVideoWindow();
}
}
}
void BlabbleCall::OnCallMediaState()
{
if (call_id_ == INVALID_CALL)
return;
pjsua_call_info info;
pj_status_t status;
if ((status = pjsua_call_get_info(call_id_, &info)) != PJ_SUCCESS) {
BLABBLE_LOG_ERROR("Unable to get call info. PJSIP call id: " << call_id_ << ", global id: " << id_ << ", pjsua_call_get_info returned " << status);
StopRinging();
return;
}
if (info.media_status == PJSUA_CALL_MEDIA_ACTIVE)
{
StopRinging();
// When media is active, connect call to sound device.
pjsua_conf_connect(info.conf_slot, 0);
pjsua_conf_connect(0, info.conf_slot);
}
}
static void on_call_media_state(pjsua_call_id call_id) {
ics_t *data;
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
//! Khoi tao mot connect media flow tu app nay sang app khac
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
pjsua_conf_connect(ci.conf_slot, 0); // ntt1->quy
pjsua_conf_connect(0, ci.conf_slot); // quy->ntt1
}
pjsua_call_get_info(call_id, &ci);
data = (ics_t *)pjsua_acc_get_user_data(ci.acc_id);
opool_item_t *p_item = opool_get(&data->opool);
build_call_media_state_event((ics_event_t *)p_item->data, call_id, ci.media_status);
ics_event_t *event = (ics_event_t *)p_item->data;
// queue_enqueue(&data->queue, (void *)p_item);
process_event(event);
}
static void _ics_conference_call(ics_t *data, int call_id) {
int i, max;
PJ_UNUSED_ARG(data);
max = pjsua_call_get_count();
SHOW_LOG(3, "Let's conference call!\n");
pjsua_call_info ci;
#if 1
if ( (call_id != current_call) && pjsua_call_is_active(call_id) ) {
pjsua_call_reinvite(call_id, PJ_TRUE, NULL);
for (i = 0; i < max; i++) {
if (pjsua_call_has_media(i) != 0) {
pjsua_conf_connect(pjsua_call_get_conf_port(call_id), pjsua_call_get_conf_port(i));
pjsua_conf_connect(pjsua_call_get_conf_port(i), pjsua_call_get_conf_port(call_id));
}
}
}
else
SHOW_LOG(3, "Cannot transfer call!\n");
#endif
//For test only:
#if 1
for (i = 0; i < max; i++){
if (pjsua_call_is_active(i) && (i != current_call)) {
pjsua_call_reinvite(i, PJ_TRUE, NULL);
pjsua_call_get_info(i, &ci);
pjsua_conf_connect(pjsua_call_get_conf_port(ci.id), pjsua_call_get_conf_port(current_call));
pjsua_conf_connect(pjsua_call_get_conf_port(current_call), pjsua_call_get_conf_port(ci.id));
}
break;
}
#endif
}
void QVDoorcom::onCallState(int call_id, QString state) {
if (call_id != active_call) {
return;
}
qDebug() << "new state" << state;
if (state == "DISCONNCTD") {
popup->hide();
active_call = -1;
// pjsua_conf_adjust_tx_level(0,0.0);
}
if (state == "CONFIRMED") {
w_dooropen->show();
/* Create DTMF generator */
pjsua_call_info ci;
qDebug() << "2" << pjmedia_tonegen_create(pj_pool, 16000, 1, 160, 16, 0, &pj_tonegen);
qDebug() << "3" << pjsua_conf_add_port(pj_pool, pj_tonegen, &pj_toneslot);
pjsua_call_get_info(call_id, &ci);
qDebug() << "4" << pjsua_conf_connect(pj_toneslot, ci.conf_slot);
/* Send DTMF code */
if (!code_accept.isEmpty()) {
qDebug() << "accept code" << code_accept;
pjmedia_tone_digit d[16];
unsigned i, count = code_accept.length();
if (count > PJ_ARRAY_SIZE(d)) {
count = PJ_ARRAY_SIZE(d);
}
pj_bzero(d, sizeof(d));
for (i=0; i<count; i++) {
d[i].digit = code_accept.at(i).toLatin1();
d[i].on_msec = 80;
d[i].off_msec = 80;
d[i].volume = 32767;
}
qDebug() << "before tonegen";
qDebug() << "tonegen result " << pjmedia_tonegen_play_digits(pj_tonegen, count, d, 0);
dtmf_timer.start();
}
}
}
// helper for creating call-recorder
static void create_recorder(pjsua_call_info ci)
{
// specify target file
pj_str_t rec_file = pj_str(app_cfg.record_file);
pj_status_t status = PJ_ENOTFOUND;
log_message("Creating recorder ... ");
// Create recorder for call
status = pjsua_recorder_create(&rec_file, 0, NULL, 0, 0, &rec_id);
if (status != PJ_SUCCESS) error_exit("Error recording answer", status);
// connect active call to call recorder
pjsua_conf_port_id rec_port = pjsua_recorder_get_conf_port(rec_id);
pjsua_conf_connect(ci.conf_slot, rec_port);
log_message("Done.\n");
}
/*****************************************************************************
* test: play simple ringback tone and hear it
*/
static void systest_play_tone(void)
{
/* Ringtones */
#define RINGBACK_FREQ1 440 /* 400 */
#define RINGBACK_FREQ2 480 /* 450 */
#define RINGBACK_ON 3000 /* 400 */
#define RINGBACK_OFF 4000 /* 200 */
#define RINGBACK_CNT 1 /* 2 */
#define RINGBACK_INTERVAL 4000 /* 2000 */
unsigned i, samples_per_frame;
pjmedia_tone_desc tone[RINGBACK_CNT];
pj_pool_t *pool = NULL;
pjmedia_port *ringback_port = NULL;
enum gui_key key;
int ringback_slot = -1;
test_item_t *ti;
pj_str_t name;
const char *title = "Audio Tone Playback Test";
pj_status_t status;
ti = systest_alloc_test_item(title);
if (!ti)
return;
key = gui_msgbox(title,
"This test will play simple ringback tone to "
"the speaker. Please listen carefully for audio "
"impairments such as stutter. You may need "
"to let this test running for a while to "
"make sure that everything is okay. Press "
"OK to start, CANCEL to skip",
WITH_OKCANCEL);
if (key != KEY_OK) {
ti->skipped = PJ_TRUE;
return;
}
PJ_LOG(3,(THIS_FILE, "Running %s", title));
pool = pjsua_pool_create("ringback", 512, 512);
samples_per_frame = systest.media_cfg.audio_frame_ptime *
systest.media_cfg.clock_rate *
systest.media_cfg.channel_count / 1000;
/* Ringback tone (call is ringing) */
name = pj_str("ringback");
status = pjmedia_tonegen_create2(pool, &name,
systest.media_cfg.clock_rate,
systest.media_cfg.channel_count,
samples_per_frame,
16, PJMEDIA_TONEGEN_LOOP,
&ringback_port);
if (status != PJ_SUCCESS)
goto on_return;
pj_bzero(&tone, sizeof(tone));
for (i=0; i<RINGBACK_CNT; ++i) {
tone[i].freq1 = RINGBACK_FREQ1;
tone[i].freq2 = RINGBACK_FREQ2;
tone[i].on_msec = RINGBACK_ON;
tone[i].off_msec = RINGBACK_OFF;
}
tone[RINGBACK_CNT-1].off_msec = RINGBACK_INTERVAL;
status = pjmedia_tonegen_play(ringback_port, RINGBACK_CNT, tone,
PJMEDIA_TONEGEN_LOOP);
if (status != PJ_SUCCESS)
goto on_return;
status = pjsua_conf_add_port(pool, ringback_port, &ringback_slot);
if (status != PJ_SUCCESS)
goto on_return;
status = pjsua_conf_connect(ringback_slot, 0);
if (status != PJ_SUCCESS)
goto on_return;
key = gui_msgbox(title,
"Ringback tone should be playing now in the "
"speaker. Press OK to stop. ", WITH_OK);
status = PJ_SUCCESS;
on_return:
if (ringback_slot != -1)
pjsua_conf_remove_port(ringback_slot);
if (ringback_port)
pjmedia_port_destroy(ringback_port);
if (pool)
pj_pool_release(pool);
if (status != PJ_SUCCESS) {
systest_perror("Sorry we encounter error when initializing "
"the tone generator: ", status);
ti->success = PJ_FALSE;
pj_strerror(status, ti->reason, sizeof(ti->reason));
} else {
key = gui_msgbox(title, "Is the audio okay?", WITH_YESNO);
ti->success = (key == KEY_YES);
if (!ti->success)
pj_ansi_strcpy(ti->reason, USER_ERROR);
}
return;
}
int dll_playWav(char* wavFile, int callId)
{
pj_status_t status;
/* Infos Player */
pjsua_player_id player_id; // Ident. for the player
pjmedia_port *media_port; // Struct. media_port
pjsua_conf_port_id conf_port; // Conference port for the player
/* Infos Call Session */
pjsua_call_info call_info;
/********************************************
/ 1- Check if Call exists and is active
********************************************/
// Get call_info from callId
pjsua_call_get_info(callId, &call_info);
if (call_info.media_status != PJSUA_CALL_MEDIA_ACTIVE)
return -1;
/********************************************
/ 2- Load the WAV File - Create the player
********************************************/
status = pjsua_player_create(&pj_str(wavFile), PJMEDIA_FILE_NO_LOOP, &player_id);
/********************************************
/ 3- Register the Callback C++ Function "on_wavplayerEof_callback"
********************************************/
if (status == PJ_SUCCESS)
{
// Get media_port from player_id
status = pjsua_player_get_port(player_id, &media_port);
}
if (status == PJ_SUCCESS)
{
// Prepare argument for Callback
wavplayerEof_Data* args = (wavplayerEof_Data*)malloc(sizeof(wavplayerEof_Data));
args->playerId = player_id;
args->callId = callId;
// Register the Callback, launched when the End of the Wave File is reached
status = pjmedia_wav_player_set_eof_cb(media_port, args, &on_wavplayerEof_callback);
}
/********************************************
/ 4- Stream the file to the Call Session
********************************************/
// Get conf_port from player_id
if (status == PJ_SUCCESS)
conf_port = pjsua_player_get_conf_port(player_id);
// one way connect conf_port (wav player) to call_info.conf_slot (call)
if ((status == PJ_SUCCESS)&&(conf_port != PJSUA_INVALID_ID)&&(call_info.conf_slot != 0)) // test if conf_port valid, and if conf_slot != soundcard
{
status = pjsua_conf_connect(conf_port, call_info.conf_slot);
}
PJ_LOG(3,(THIS_FILE,"Wav Play, status %d",status));
if (status != PJ_SUCCESS)
return -1;
return player_id;
}
/*
* Callback on media state changed event.
* The action may connect the call to sound device, to file, or
* to loop the call.
*/
static void on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info call_info;
pjsua_call_get_info(call_id, &call_info);
/* Connect ports appropriately when media status is ACTIVE or REMOTE HOLD,
* otherwise we should NOT connect the ports.
*/
if (call_info.media_status == PJSUA_CALL_MEDIA_ACTIVE ||
call_info.media_status == PJSUA_CALL_MEDIA_REMOTE_HOLD)
{
pj_bool_t connect_sound = PJ_TRUE;
/* Loopback sound, if desired */
if (app_config.auto_loop) {
pjsua_conf_connect(call_info.conf_slot, call_info.conf_slot);
connect_sound = PJ_FALSE;
}
/* Automatically record conversation, if desired */
if (app_config.auto_rec && app_config.rec_port != PJSUA_INVALID_ID) {
pjsua_conf_connect(call_info.conf_slot, app_config.rec_port);
}
/* Stream a file, if desired */
if ((app_config.auto_play || app_config.auto_play_hangup) &&
app_config.wav_port != PJSUA_INVALID_ID)
{
pjsua_conf_connect(app_config.wav_port, call_info.conf_slot);
connect_sound = PJ_FALSE;
}
/* Put call in conference with other calls, if desired */
if (app_config.auto_conf) {
pjsua_call_id call_ids[PJSUA_MAX_CALLS];
unsigned call_cnt=PJ_ARRAY_SIZE(call_ids);
unsigned i;
/* Get all calls, and establish media connection between
* this call and other calls.
*/
pjsua_enum_calls(call_ids, &call_cnt);
for (i=0; i<call_cnt; ++i) {
if (call_ids[i] == call_id)
continue;
if (!pjsua_call_has_media(call_ids[i]))
continue;
pjsua_conf_connect(call_info.conf_slot,
pjsua_call_get_conf_port(call_ids[i]));
pjsua_conf_connect(pjsua_call_get_conf_port(call_ids[i]),
call_info.conf_slot);
/* Automatically record conversation, if desired */
if (app_config.auto_rec && app_config.rec_port != PJSUA_INVALID_ID) {
pjsua_conf_connect(pjsua_call_get_conf_port(call_ids[i]),
app_config.rec_port);
}
}
/* Also connect call to local sound device */
connect_sound = PJ_TRUE;
}
/* Otherwise connect to sound device */
if (connect_sound) {
pjsua_conf_connect(call_info.conf_slot, 0);
pjsua_conf_connect(0, call_info.conf_slot);
/* Automatically record conversation, if desired */
if (app_config.auto_rec && app_config.rec_port != PJSUA_INVALID_ID) {
pjsua_conf_connect(call_info.conf_slot, app_config.rec_port);
pjsua_conf_connect(0, app_config.rec_port);
}
}
}
PJ_LOG(3,(THIS_FILE, "Media for call %d is active", call_id));
/* Handle media status */
switch (call_info.media_status) {
case PJSUA_CALL_MEDIA_ACTIVE:
PJ_LOG(3,(THIS_FILE, "Media for call %d is active", call_id));
break;
case PJSUA_CALL_MEDIA_LOCAL_HOLD:
PJ_LOG(3,(THIS_FILE, "Media for call %d is suspended (hold) by local",
call_id));
// call on call hold handler
cb_callholdconf(call_id);
break;
case PJSUA_CALL_MEDIA_REMOTE_HOLD:
PJ_LOG(3,(THIS_FILE,
"Media for call %d is suspended (hold) by remote",
call_id));
break;
case PJSUA_CALL_MEDIA_ERROR:
PJ_LOG(3,(THIS_FILE,
"Media has reported error, disconnecting call"));
{
pj_str_t reason = pj_str("ICE negotiation failed");
pjsua_call_hangup(call_id, 500, &reason, NULL);
}
break;
case PJSUA_CALL_MEDIA_NONE:
PJ_LOG(3,(THIS_FILE,
"Media for call %d is inactive",
call_id));
break;
default:
pj_assert(!"Unhandled media status");
break;
}
}
static void systest_aec_test(void)
{
const char *ref_wav_paths[] = { add_path(res_path, WAV_PLAYBACK_PATH),
ALT_PATH1 WAV_PLAYBACK_PATH };
pjsua_player_id player_id = PJSUA_INVALID_ID;
pjsua_recorder_id writer_id = PJSUA_INVALID_ID;
enum gui_key key;
test_item_t *ti;
const char *title = "AEC/AES Test";
unsigned last_ec_tail = 0;
pj_status_t status;
pj_str_t tmp;
ti = systest_alloc_test_item(title);
if (!ti)
return;
key = gui_msgbox(title,
"This test will try to find whether the AEC/AES "
"works good on this system. Test will play a file "
"while recording from mic. The recording will be "
"played back later so you can check if echo is there. "
"Press OK to start.",
WITH_OKCANCEL);
if (key != KEY_OK) {
ti->skipped = PJ_TRUE;
return;
}
/* Save current EC tail */
status = pjsua_get_ec_tail(&last_ec_tail);
if (status != PJ_SUCCESS)
goto on_return;
/* Set EC tail setting to default */
status = pjsua_set_ec(PJSUA_DEFAULT_EC_TAIL_LEN, 0);
if (status != PJ_SUCCESS)
goto on_return;
/*
* Create player and recorder
*/
status = create_player(PJ_ARRAY_SIZE(ref_wav_paths), ref_wav_paths,
&player_id);
if (status != PJ_SUCCESS) {
PJ_PERROR(1,(THIS_FILE, status, "Error opening WAV file %s",
WAV_PLAYBACK_PATH));
goto on_return;
}
status = pjsua_recorder_create(
pj_cstr(&tmp, add_path(doc_path, AEC_REC_PATH)), 0, 0, -1,
0, &writer_id);
if (status != PJ_SUCCESS) {
PJ_PERROR(1,(THIS_FILE, status, "Error writing WAV file %s",
AEC_REC_PATH));
goto on_return;
}
/*
* Start playback and recording.
*/
pjsua_conf_connect(pjsua_player_get_conf_port(player_id), 0);
pj_thread_sleep(100);
pjsua_conf_connect(0, pjsua_recorder_get_conf_port(writer_id));
/* Wait user signal */
gui_msgbox(title, "AEC/AES test is running. Press OK to stop this test.",
WITH_OK);
/*
* Stop and close playback and recorder
*/
pjsua_conf_disconnect(0, pjsua_recorder_get_conf_port(writer_id));
pjsua_conf_disconnect(pjsua_player_get_conf_port(player_id), 0);
pjsua_recorder_destroy(writer_id);
pjsua_player_destroy(player_id);
player_id = PJSUA_INVALID_ID;
writer_id = PJSUA_INVALID_ID;
/*
* Play the result.
*/
status = pjsua_player_create(
pj_cstr(&tmp, add_path(doc_path, AEC_REC_PATH)),
0, &player_id);
if (status != PJ_SUCCESS) {
PJ_PERROR(1,(THIS_FILE, status, "Error opening WAV file %s", AEC_REC_PATH));
goto on_return;
}
pjsua_conf_connect(pjsua_player_get_conf_port(player_id), 0);
/* Wait user signal */
gui_msgbox(title, "We are now playing the captured audio from the mic. "
"Check if echo (of the audio played back previously) is "
"present in the audio. The recording is stored in "
AEC_REC_PATH " for offline analysis. "
"Press OK to stop.",
WITH_OK);
pjsua_conf_disconnect(pjsua_player_get_conf_port(player_id), 0);
key = gui_msgbox(title,
"Did you notice any echo in the recording?",
WITH_YESNO);
on_return:
if (player_id != PJSUA_INVALID_ID)
pjsua_player_destroy(player_id);
if (writer_id != PJSUA_INVALID_ID)
pjsua_recorder_destroy(writer_id);
/* Wait until sound device closed before restoring back EC tail setting */
while (pjsua_snd_is_active())
pj_thread_sleep(10);
pjsua_set_ec(last_ec_tail, 0);
if (status != PJ_SUCCESS) {
systest_perror("Sorry we encountered an error: ", status);
ti->success = PJ_FALSE;
pj_strerror(status, ti->reason, sizeof(ti->reason));
} else if (key == KEY_YES) {
ti->success = PJ_FALSE;
if (!ti->success) {
pj_ansi_strcpy(ti->reason, USER_ERROR);
}
} else {
char msg[200];
pj_ansi_snprintf(msg, sizeof(msg), "Test succeeded.\r\n");
ti->success = PJ_TRUE;
pj_ansi_strncpy(ti->reason, msg, sizeof(ti->reason));
ti->reason[sizeof(ti->reason)-1] = '\0';
}
}
static void systest_latency_test(void)
{
const char *ref_wav_paths[] = { add_path(res_path, WAV_TOCK8_PATH), ALT_PATH1 WAV_TOCK8_PATH };
pj_str_t rec_wav_file;
pjsua_player_id play_id = PJSUA_INVALID_ID;
pjsua_conf_port_id play_slot = PJSUA_INVALID_ID;
pjsua_recorder_id rec_id = PJSUA_INVALID_ID;
pjsua_conf_port_id rec_slot = PJSUA_INVALID_ID;
pj_pool_t *pool = NULL;
pjmedia_port *wav_port = NULL;
unsigned lat_sum=0, lat_cnt=0, lat_min=0, lat_max=0;
enum gui_key key;
test_item_t *ti;
const char *title = "Audio Latency Test";
pj_status_t status;
ti = systest_alloc_test_item(title);
if (!ti)
return;
key = gui_msgbox(title,
"This test will try to find the audio device's "
"latency. We will play a special WAV file to the "
"speaker for ten seconds, then at the end "
"calculate the latency. Please don't do anything "
"until the test is done.", WITH_OKCANCEL);
if (key != KEY_OK) {
ti->skipped = PJ_TRUE;
return;
}
key = gui_msgbox(title,
"For this test to work, we must be able to capture "
"the audio played in the speaker (the echo), and only"
" that audio (i.e. you must be in relatively quiet "
"place to run this test). "
"Press OK to start, or CANCEL to skip.",
WITH_OKCANCEL);
if (key != KEY_OK) {
ti->skipped = PJ_TRUE;
return;
}
PJ_LOG(3,(THIS_FILE, "Running %s", title));
status = create_player(PJ_ARRAY_SIZE(ref_wav_paths), ref_wav_paths,
&play_id);
if (status != PJ_SUCCESS)
goto on_return;
play_slot = pjsua_player_get_conf_port(play_id);
rec_wav_file = pj_str(add_path(doc_path, WAV_LATENCY_OUT_PATH));
status = pjsua_recorder_create(&rec_wav_file, 0, NULL, -1, 0, &rec_id);
if (status != PJ_SUCCESS)
goto on_return;
rec_slot = pjsua_recorder_get_conf_port(rec_id);
/* Setup the test */
//status = pjsua_conf_connect(0, 0);
status = pjsua_conf_connect(play_slot, 0);
status = pjsua_conf_connect(0, rec_slot);
status = pjsua_conf_connect(play_slot, rec_slot);
/* We're running */
PJ_LOG(3,(THIS_FILE, "Please wait while test is running (~10 sec)"));
gui_sleep(10);
/* Done with the test */
//status = pjsua_conf_disconnect(0, 0);
status = pjsua_conf_disconnect(play_slot, rec_slot);
status = pjsua_conf_disconnect(0, rec_slot);
status = pjsua_conf_disconnect(play_slot, 0);
pjsua_recorder_destroy(rec_id);
rec_id = PJSUA_INVALID_ID;
pjsua_player_destroy(play_id);
play_id = PJSUA_INVALID_ID;
/* Confirm that echo is heard */
gui_msgbox(title,
"Test is done. Now we need to confirm that we indeed "
"captured the echo. We will play the captured audio "
"and please confirm that you can hear the 'tock' echo.",
WITH_OK);
status = pjsua_player_create(&rec_wav_file, 0, &play_id);
if (status != PJ_SUCCESS)
goto on_return;
play_slot = pjsua_player_get_conf_port(play_id);
status = pjsua_conf_connect(play_slot, 0);
if (status != PJ_SUCCESS)
goto on_return;
key = gui_msgbox(title,
"The captured audio is being played back now. "
"Can you hear the 'tock' echo?",
WITH_YESNO);
pjsua_player_destroy(play_id);
play_id = PJSUA_INVALID_ID;
if (key != KEY_YES)
goto on_return;
/* Now analyze the latency */
pool = pjsua_pool_create("latency", 512, 512);
status = pjmedia_wav_player_port_create(pool, rec_wav_file.ptr, 0, 0, 0, &wav_port);
if (status != PJ_SUCCESS)
goto on_return;
status = calculate_latency(pool, wav_port, &lat_sum, &lat_cnt,
&lat_min, &lat_max);
if (status != PJ_SUCCESS)
goto on_return;
on_return:
if (wav_port)
pjmedia_port_destroy(wav_port);
if (pool)
pj_pool_release(pool);
if (play_id != PJSUA_INVALID_ID)
pjsua_player_destroy(play_id);
if (rec_id != PJSUA_INVALID_ID)
pjsua_recorder_destroy(rec_id);
if (status != PJ_SUCCESS) {
systest_perror("Sorry we encountered an error: ", status);
ti->success = PJ_FALSE;
pj_strerror(status, ti->reason, sizeof(ti->reason));
} else if (key != KEY_YES) {
ti->success = PJ_FALSE;
if (!ti->success) {
pj_ansi_strcpy(ti->reason, USER_ERROR);
}
} else {
char msg[200];
pj_size_t msglen;
pj_ansi_snprintf(msg, sizeof(msg),
"The sound device latency:\r\n"
" Min=%u, Max=%u, Avg=%u\r\n",
lat_min, lat_max, lat_sum/lat_cnt);
msglen = strlen(msg);
if (lat_sum/lat_cnt > 500) {
pj_ansi_snprintf(msg+msglen, sizeof(msg)-msglen,
"The latency is huge!\r\n");
msglen = strlen(msg);
} else if (lat_sum/lat_cnt > 200) {
pj_ansi_snprintf(msg+msglen, sizeof(msg)-msglen,
"The latency is quite high\r\n");
msglen = strlen(msg);
}
key = gui_msgbox(title, msg, WITH_OK);
ti->success = PJ_TRUE;
pj_ansi_strncpy(ti->reason, msg, sizeof(ti->reason));
ti->reason[sizeof(ti->reason)-1] = '\0';
}
}
/*****************************************************************************
* test: record audio
*/
static void systest_rec_audio(void)
{
const pj_str_t filename = pj_str(add_path(doc_path, WAV_REC_OUT_PATH));
pj_pool_t *pool = NULL;
enum gui_key key;
pjsua_recorder_id rec_id = PJSUA_INVALID_ID;
pjsua_player_id play_id = PJSUA_INVALID_ID;
pjsua_conf_port_id rec_slot = PJSUA_INVALID_ID;
pjsua_conf_port_id play_slot = PJSUA_INVALID_ID;
pj_status_t status = PJ_SUCCESS;
const char *title = "Audio Recording";
test_item_t *ti;
ti = systest_alloc_test_item(title);
if (!ti)
return;
key = gui_msgbox(title,
"This test will allow you to record audio "
"from the microphone, and playback the "
"audio to the speaker. Press OK to start recording, "
"CANCEL to skip.",
WITH_OKCANCEL);
if (key != KEY_OK) {
ti->skipped = PJ_TRUE;
return;
}
PJ_LOG(3,(THIS_FILE, "Running %s", title));
pool = pjsua_pool_create("rectest", 512, 512);
status = pjsua_recorder_create(&filename, 0, NULL, -1, 0, &rec_id);
if (status != PJ_SUCCESS)
goto on_return;
rec_slot = pjsua_recorder_get_conf_port(rec_id);
status = pjsua_conf_connect(0, rec_slot);
if (status != PJ_SUCCESS)
goto on_return;
key = gui_msgbox(title,
"Recording is in progress now, please say "
"something in the microphone. Press OK "
"to stop recording", WITH_OK);
pjsua_conf_disconnect(0, rec_slot);
rec_slot = PJSUA_INVALID_ID;
pjsua_recorder_destroy(rec_id);
rec_id = PJSUA_INVALID_ID;
status = pjsua_player_create(&filename, 0, &play_id);
if (status != PJ_SUCCESS)
goto on_return;
play_slot = pjsua_player_get_conf_port(play_id);
status = pjsua_conf_connect(play_slot, 0);
if (status != PJ_SUCCESS)
goto on_return;
key = gui_msgbox(title,
"Recording has been stopped. "
"The recorded audio is being played now to "
"the speaker device, in a loop. Listen for "
"any audio impairments. Press OK to stop.",
WITH_OK);
on_return:
if (rec_slot != PJSUA_INVALID_ID)
pjsua_conf_disconnect(0, rec_slot);
if (rec_id != PJSUA_INVALID_ID)
pjsua_recorder_destroy(rec_id);
if (play_slot != PJSUA_INVALID_ID)
pjsua_conf_disconnect(play_slot, 0);
if (play_id != PJSUA_INVALID_ID)
pjsua_player_destroy(play_id);
if (pool)
pj_pool_release(pool);
if (status != PJ_SUCCESS) {
systest_perror("Sorry we encountered an error: ", status);
ti->success = PJ_FALSE;
pj_strerror(status, ti->reason, sizeof(ti->reason));
} else {
key = gui_msgbox(title, "Is the audio okay?", WITH_YESNO);
ti->success = (key == KEY_YES);
if (!ti->success) {
pj_ansi_snprintf(textbuf, sizeof(textbuf),
"You will probably need to copy the recorded "
"WAV file %s to a desktop computer and analyze "
"it, to find out whether it's a recording "
"or playback problem.",
WAV_REC_OUT_PATH);
gui_msgbox(title, textbuf, WITH_OK);
pj_ansi_strcpy(ti->reason, USER_ERROR);
}
}
}
