static void create_text_stream(text_stream_tester_t *tst, int payload_type) {
tst->ts = text_stream_new2(tst->local_ip, tst->local_rtp, tst->local_rtcp);
tst->local_rtp = rtp_session_get_local_port(tst->ts->ms.sessions.rtp_session);
tst->local_rtcp = rtp_session_get_local_rtcp_port(tst->ts->ms.sessions.rtp_session);
reset_stats(&tst->stats);
rtp_session_set_multicast_loopback(tst->ts->ms.sessions.rtp_session, TRUE);
tst->stats.q = ortp_ev_queue_new();
rtp_session_register_event_queue(tst->ts->ms.sessions.rtp_session, tst->stats.q);
tst->payload_type = payload_type;
}
// Constructors
MastTool::MastTool( const char* tool_name, RtpSessionMode mode )
{
int log_level = ORTP_WARNING|ORTP_ERROR|ORTP_FATAL;
// Initialise defaults
this->session = NULL;
this->profile = &av_profile;
this->mimetype = new MastMimeType();
this->payloadtype = NULL;
this->payloadtype_index = -1;
this->tool_name = tool_name;
this->payload_size_limit = DEFAULT_PAYLOAD_LIMIT;
// Initialise the oRTP library
ortp_init();
// Set the logging message level
#ifdef DEBUGGING
MAST_DEBUG( "Compiled with debugging enabled" );
log_level |= ORTP_DEBUG;
log_level |= ORTP_MESSAGE;
#endif
ortp_set_log_level_mask(log_level);
// Create RTP session
session = rtp_session_new( mode );
if (session==NULL) {
MAST_FATAL( "Failed to create oRTP session.\n" );
}
// Enabled multicast loopback
rtp_session_set_multicast_loopback(session, TRUE);
// Callbacks
rtp_session_signal_connect(session,"ssrc_changed",(RtpCallback)ssrc_changed_cb, 0);
rtp_session_signal_connect(session,"payload_type_changed",(RtpCallback)pt_changed_cb, 0);
rtp_session_signal_connect(session,"network_error",(RtpCallback)network_error_cb, 0);
// Set the MPEG Audio payload type to 14 in the AV profile
rtp_profile_set_payload(profile, RTP_MPEG_AUDIO_PT, &payload_type_mpeg_audio);
// Set RTCP parameters
this->set_source_sdes();
}
RtpSession * ms_create_duplex_rtp_session(const char* local_ip, int loc_rtp_port, int loc_rtcp_port, int mtu) {
RtpSession *rtpr;
rtpr = rtp_session_new(RTP_SESSION_SENDRECV);
rtp_session_set_recv_buf_size(rtpr, MAX(mtu , MS_MINIMAL_MTU));
rtp_session_set_scheduling_mode(rtpr, 0);
rtp_session_set_blocking_mode(rtpr, 0);
rtp_session_enable_adaptive_jitter_compensation(rtpr, TRUE);
rtp_session_set_symmetric_rtp(rtpr, TRUE);
rtp_session_set_local_addr(rtpr, local_ip, loc_rtp_port, loc_rtcp_port);
rtp_session_signal_connect(rtpr, "timestamp_jump", (RtpCallback)rtp_session_resync, NULL);
rtp_session_signal_connect(rtpr, "ssrc_changed", (RtpCallback)rtp_session_resync, NULL);
rtp_session_set_ssrc_changed_threshold(rtpr, 0);
rtp_session_set_rtcp_report_interval(rtpr, 2500); /* At the beginning of the session send more reports. */
rtp_session_set_multicast_loopback(rtpr,TRUE); /*very useful, specially for testing purposes*/
disable_checksums(rtp_session_get_rtp_socket(rtpr));
return rtpr;
}
static void create_video_stream(video_stream_tester_t *vst, int payload_type) {
vst->vs = video_stream_new2(vst->local_ip, vst->local_rtp, vst->local_rtcp);
vst->vs->staticimage_webcam_fps_optimization = FALSE;
vst->local_rtp = rtp_session_get_local_port(vst->vs->ms.sessions.rtp_session);
vst->local_rtcp = rtp_session_get_local_rtcp_port(vst->vs->ms.sessions.rtp_session);
reset_stats(&vst->stats);
rtp_session_set_multicast_loopback(vst->vs->ms.sessions.rtp_session, TRUE);
vst->stats.q = ortp_ev_queue_new();
rtp_session_register_event_queue(vst->vs->ms.sessions.rtp_session, vst->stats.q);
video_stream_set_event_callback(vst->vs, video_stream_event_cb, vst);
if (vst->vconf) {
PayloadType *pt = rtp_profile_get_payload(&rtp_profile, payload_type);
CU_ASSERT_PTR_NOT_NULL_FATAL(pt);
pt->normal_bitrate = vst->vconf->required_bitrate;
video_stream_set_fps(vst->vs, vst->vconf->fps);
video_stream_set_sent_video_size(vst->vs, vst->vconf->vsize);
}
vst->payload_type = payload_type;
}
static void basic_audio_stream_base( const char* marielle_local_ip
, int marielle_local_rtp_port
, int marielle_local_rtcp_port
, const char* margaux_local_ip
, int margaux_local_rtp_port
, int margaux_local_rtcp_port) {
AudioStream * marielle = audio_stream_new2 (marielle_local_ip, marielle_local_rtp_port, marielle_local_rtcp_port);
stats_t marielle_stats;
AudioStream * margaux = audio_stream_new2 (margaux_local_ip, margaux_local_rtp_port,margaux_local_rtcp_port);
stats_t margaux_stats;
RtpProfile* profile = rtp_profile_new("default profile");
char* hello_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_file_root(), HELLO_8K_1S_FILE);
char* recorded_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_writable_dir(), RECORDED_8K_1S_FILE);
int dummy=0;
rtp_session_set_multicast_loopback(marielle->ms.sessions.rtp_session,TRUE);
rtp_session_set_multicast_loopback(margaux->ms.sessions.rtp_session,TRUE);
reset_stats(&marielle_stats);
reset_stats(&margaux_stats);
rtp_profile_set_payload (profile,0,&payload_type_pcmu8000);
CU_ASSERT_EQUAL(audio_stream_start_full(margaux
, profile
, ms_is_multicast(margaux_local_ip)?margaux_local_ip:marielle_local_ip
, ms_is_multicast(margaux_local_ip)?margaux_local_rtp_port:marielle_local_rtp_port
, marielle_local_ip
, marielle_local_rtcp_port
, 0
, 50
, NULL
, recorded_file
, NULL
, NULL
, 0),0);
CU_ASSERT_EQUAL(audio_stream_start_full(marielle
, profile
, margaux_local_ip
, margaux_local_rtp_port
, margaux_local_ip
, margaux_local_rtcp_port
, 0
, 50
, hello_file
, NULL
, NULL
, NULL
, 0),0);
ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);
CU_ASSERT_TRUE(wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,1,12000));
/*make sure packets can cross from sender to receiver*/
wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,500);
audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);
/* No packet loss is assumed */
CU_ASSERT_EQUAL(marielle_stats.rtp.sent,margaux_stats.rtp.recv);
audio_stream_stop(marielle);
audio_stream_stop(margaux);
unlink(recorded_file);
ms_free(recorded_file);
ms_free(hello_file);
}
static void basic_audio_stream_base_2( const char* marielle_local_ip
, const char* marielle_remote_ip
, int marielle_local_rtp_port
, int marielle_remote_rtp_port
, int marielle_local_rtcp_port
, int marielle_remote_rtcp_port
, const char* margaux_local_ip
, const char* margaux_remote_ip
, int margaux_local_rtp_port
, int margaux_remote_rtp_port
, int margaux_local_rtcp_port
, int margaux_remote_rtcp_port
, int lost_percentage) {
AudioStream * marielle = audio_stream_new2 (_factory, marielle_local_ip, marielle_local_rtp_port, marielle_local_rtcp_port);
stats_t marielle_stats;
AudioStream * margaux = audio_stream_new2 (_factory, margaux_local_ip, margaux_local_rtp_port,margaux_local_rtcp_port);
stats_t margaux_stats;
RtpProfile* profile = rtp_profile_new("default profile");
char* hello_file = bc_tester_res(HELLO_8K_1S_FILE);
char* recorded_file = bc_tester_file(RECORDED_8K_1S_FILE);
uint64_t marielle_rtp_sent=0;
rtp_session_set_multicast_loopback(marielle->ms.sessions.rtp_session,TRUE);
rtp_session_set_multicast_loopback(margaux->ms.sessions.rtp_session,TRUE);
rtp_session_set_rtcp_report_interval(marielle->ms.sessions.rtp_session, 1000);
rtp_session_set_rtcp_report_interval(margaux->ms.sessions.rtp_session, 1000);
reset_stats(&marielle_stats);
reset_stats(&margaux_stats);
rtp_profile_set_payload (profile,0,&payload_type_pcmu8000);
BC_ASSERT_EQUAL(audio_stream_start_full(margaux
, profile
, ms_is_multicast(margaux_local_ip)?margaux_local_ip:margaux_remote_ip
, ms_is_multicast(margaux_local_ip)?margaux_local_rtp_port:margaux_remote_rtp_port
, margaux_remote_ip
, margaux_remote_rtcp_port
, 0
, 50
, NULL
, recorded_file
, NULL
, NULL
, 0)
,0, int, "%d");
BC_ASSERT_EQUAL(audio_stream_start_full(marielle
, profile
, marielle_remote_ip
, marielle_remote_rtp_port
, marielle_remote_ip
, marielle_remote_rtcp_port
, 0
, 50
, hello_file
, NULL
, NULL
, NULL
, 0)
,0, int, "%d");
ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);
wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,1,12000);
audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);
marielle_rtp_sent = marielle_stats.rtp.sent;
if (rtp_session_rtcp_enabled(marielle->ms.sessions.rtp_session) && rtp_session_rtcp_enabled(margaux->ms.sessions.rtp_session)) {
BC_ASSERT_GREATER_STRICT(rtp_session_get_round_trip_propagation(marielle->ms.sessions.rtp_session),0,float,"%f");
BC_ASSERT_GREATER_STRICT(rtp_session_get_stats(marielle->ms.sessions.rtp_session)->recv_rtcp_packets,0,unsigned long long,"%llu");
}
