static void basic_audio_stream() {
AudioStream * marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
stats_t marielle_stats;
AudioStream * margaux = audio_stream_new (MARGAUX_RTP_PORT,MARGAUX_RTCP_PORT, FALSE);
stats_t margaux_stats;
RtpProfile* profile = rtp_profile_new("default profile");
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
, MARIELLE_IP
, MARIELLE_RTP_PORT
, MARIELLE_IP
, MARIELLE_RTCP_PORT
, 0
, 50
, NULL
, RECORDED_8K_1S_FILE
, NULL
, NULL
, 0),0);
CU_ASSERT_EQUAL(audio_stream_start_full(marielle
, profile
, MARGAUX_IP
, MARGAUX_RTP_PORT
, MARGAUX_IP
, MARGAUX_RTCP_PORT
, 0
, 50
, HELLO_8K_1S_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));
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_8K_1S_FILE);
}
static void add_local_endpoint(LinphoneConference *conf,LinphoneCore *lc){
/*create a dummy audiostream in order to extract the local part of it */
/* network address and ports have no meaning and are not used here. */
AudioStream *st=audio_stream_new(65000,65001,FALSE);
MSSndCard *playcard=lc->sound_conf.lsd_card ?
lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
MSSndCard *captcard=lc->sound_conf.capt_sndcard;
const MSAudioConferenceParams *params=ms_audio_conference_get_params(conf->conf);
conf->local_dummy_profile=make_dummy_profile(params->samplerate);
audio_stream_start_full(st, conf->local_dummy_profile,
"127.0.0.1",
65000,
"127.0.0.1",
65001,
0,
40,
NULL,
NULL,
playcard,
captcard,
linphone_core_echo_cancellation_enabled(lc)
);
_post_configure_audio_stream(st,lc,FALSE);
conf->local_participant=st;
conf->local_endpoint=ms_audio_endpoint_get_from_stream(st,FALSE);
ms_audio_conference_add_member(conf->conf,conf->local_endpoint);
}
void linphone_gtk_start_record_sound(GtkWidget *w, gpointer data){
LinphoneCore *lc = linphone_gtk_get_core();
AudioStream *stream = NULL;
MSSndCardManager *manager = ms_snd_card_manager_get();
gboolean active=gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(w));
if(active){
gchar *path = get_record_file();
stream=audio_stream_new(8888, 8889, FALSE);
if(stream != NULL){
audio_stream_start_full(stream,&av_profile,"127.0.0.1",8888,"127.0.0.1",8889,0,0,NULL,
path,NULL,ms_snd_card_manager_get_card(manager,linphone_core_get_capture_device(lc)),FALSE);
g_object_set_data(G_OBJECT(audio_assistant),"record_stream",stream);
}
gint timeout_id = gtk_timeout_add(6000,(GtkFunction)linphone_gtk_stop_record,NULL);
g_object_set_data(G_OBJECT(audio_assistant),"timeout_id",GINT_TO_POINTER(timeout_id));
g_object_set_data(G_OBJECT(audio_assistant),"path",path);
} else {
stream = (AudioStream *)g_object_get_data(G_OBJECT(audio_assistant),"record_stream");
gint timeout_id = GPOINTER_TO_INT(g_object_get_data(G_OBJECT(audio_assistant),"timeout_id"));
gtk_timeout_remove(timeout_id);
if(stream != NULL){
audio_stream_stop(stream);
g_object_set_data(G_OBJECT(audio_assistant),"record_stream",NULL);
}
update_record_button(FALSE);
update_play_button(TRUE);
}
}
AudioStream * audio_stream_start(RtpProfile *prof,int locport,const char *remip,int remport,int profile,int jitt_comp,bool_t use_ec)
{
MSSndCard *sndcard_playback;
MSSndCard *sndcard_capture;
AudioStream *stream;
sndcard_capture=ms_snd_card_manager_get_default_capture_card(ms_snd_card_manager_get());
sndcard_playback=ms_snd_card_manager_get_default_playback_card(ms_snd_card_manager_get());
if (sndcard_capture==NULL || sndcard_playback==NULL)
return NULL;
stream=audio_stream_new(locport, locport+1, ms_is_ipv6(remip));
if (audio_stream_start_full(stream,prof,remip,remport,remip,remport+1,profile,jitt_comp,NULL,NULL,sndcard_playback,sndcard_capture,use_ec)==0) return stream;
audio_stream_free(stream);
return NULL;
}
AudioStream *audio_stream_start_with_sndcards(RtpProfile *prof,int locport,const char *remip,int remport,int profile,int jitt_comp,MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec)
{
AudioStream *stream;
if (playcard==NULL) {
ms_error("No playback card.");
return NULL;
}
if (captcard==NULL) {
ms_error("No capture card.");
return NULL;
}
stream=audio_stream_new(locport, locport+1, ms_is_ipv6(remip));
if (audio_stream_start_full(stream,prof,remip,remport,remip,remport+1,profile,jitt_comp,NULL,NULL,playcard,captcard,use_ec)==0) return stream;
audio_stream_free(stream);
return NULL;
}
static void audio_manager_start(stream_manager_t * mgr
,int payload_type
,int remote_port
,int target_bitrate
,const char* player_file
,const char* recorder_file){
media_stream_set_target_network_bitrate(&mgr->audio_stream->ms,target_bitrate);
CU_ASSERT_EQUAL(audio_stream_start_full(mgr->audio_stream
, &rtp_profile
, "127.0.0.1"
, remote_port
, "127.0.0.1"
, remote_port+1
, payload_type
, 50
, player_file
, recorder_file
, NULL
, NULL
, 0),0);
}
void linphone_gtk_start_play_record_sound(GtkWidget *w,gpointer data){
LinphoneCore *lc = linphone_gtk_get_core();
gboolean active=gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(w));
AudioStream *stream = NULL;
MSSndCardManager *manager = ms_snd_card_manager_get();
if(active){
gchar *path = g_object_get_data(G_OBJECT(audio_assistant),"path");
stream=audio_stream_new(8888, 8889, FALSE);
if(path != NULL){
audio_stream_start_full(stream,&av_profile,"127.0.0.1",8888,"127.0.0.1",8889,0,0,path,
NULL,ms_snd_card_manager_get_card(manager,linphone_core_get_playback_device(lc)),NULL,FALSE);
ms_filter_add_notify_callback(stream->soundread,endoffile_cb,stream,FALSE);
g_object_set_data(G_OBJECT(audio_assistant),"play_stream",stream);
}
} else {
stream = (AudioStream *)g_object_get_data(G_OBJECT(audio_assistant),"play_stream");
if(stream != NULL){
audio_stream_stop(stream);
g_object_set_data(G_OBJECT(audio_assistant),"play_stream",NULL);
}
}
}
int audio_stream_start_now(AudioStream *stream, RtpProfile * prof, const char *remip, int remport, int rem_rtcp_port, int payload_type, int jitt_comp, MSSndCard *playcard, MSSndCard *captcard, bool_t use_ec){
return audio_stream_start_full(stream,prof,remip,remport,remip,rem_rtcp_port,
payload_type,jitt_comp,NULL,NULL,playcard,captcard,use_ec);
}
int audio_stream_start_with_files(AudioStream *stream, RtpProfile *prof,const char *remip, int remport,
int rem_rtcp_port, int pt,int jitt_comp, const char *infile, const char * outfile)
{
return audio_stream_start_full(stream,prof,remip,remport,remip,rem_rtcp_port,pt,jitt_comp,infile,outfile,NULL,NULL,FALSE);
}
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 encrypted_audio_stream_base( bool_t change_ssrc,
bool_t change_send_key_in_the_middle
,bool_t set_both_send_recv_key
,bool_t send_key_first) {
AudioStream * marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
AudioStream * margaux = audio_stream_new (MARGAUX_RTP_PORT,MARGAUX_RTCP_PORT, FALSE);
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);
stats_t marielle_stats;
stats_t margaux_stats;
int dummy=0;
if (ms_srtp_supported()) {
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
, MARIELLE_IP
, MARIELLE_RTP_PORT
, MARIELLE_IP
, MARIELLE_RTCP_PORT
, 0
, 50
, NULL
, recorded_file
, NULL
, NULL
, 0),0);
CU_ASSERT_EQUAL(audio_stream_start_full(marielle
, profile
, MARGAUX_IP
, MARGAUX_RTP_PORT
, MARGAUX_IP
, MARGAUX_RTCP_PORT
, 0
, 50
, hello_file
, NULL
, NULL
, NULL
, 0),0);
if (send_key_first) {
CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);
if (set_both_send_recv_key)
CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") == 0);
CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") ==0);
if (set_both_send_recv_key)
CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") ==0);
} else {
CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") ==0);
if (set_both_send_recv_key)
CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") ==0);
CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);
if (set_both_send_recv_key)
CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") == 0);
}
ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);
if (change_send_key_in_the_middle) {
int dummy=0;
wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,2000);
CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "eCYF4nYyCvmCpFWjUeDaxI2GWp2BzCRlIPfg52Te") == 0);
CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "eCYF4nYyCvmCpFWjUeDaxI2GWp2BzCRlIPfg52Te") ==0);
}
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 */
if (change_send_key_in_the_middle) {
/*we can accept one or 2 error in such case*/
CU_ASSERT_TRUE((marielle_stats.rtp.packet_sent-margaux_stats.rtp.packet_recv)<3);
} else
CU_ASSERT_EQUAL(marielle_stats.rtp.sent,margaux_stats.rtp.recv);
if (change_ssrc) {
audio_stream_stop(marielle);
marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
CU_ASSERT_EQUAL(audio_stream_start_full(marielle
, profile
, MARGAUX_IP
, MARGAUX_RTP_PORT
, MARGAUX_IP
, MARGAUX_RTCP_PORT
, 0
, 50
, hello_file
, NULL
, NULL
, NULL
, 0),0);
CU_ASSERT_FATAL(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 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,2,12000));
/*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*2,margaux_stats.rtp.recv);
}
unlink(recorded_file);
ms_free(recorded_file);
ms_free(hello_file);
} else {
ms_warning("srtp not available, skiping...");
}
audio_stream_stop(marielle);
audio_stream_stop(margaux);
rtp_profile_destroy(profile);
}
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");
}
static void run_media_streams(int localport, const char *remote_ip, int remoteport, int payload, const char *fmtp,
int jitter, int bitrate, MSVideoSize vs, bool_t ec, bool_t agc, bool_t eq)
{
AudioStream *audio=NULL;
#ifdef VIDEO_ENABLED
VideoStream *video=NULL;
#endif
RtpSession *session=NULL;
PayloadType *pt;
RtpProfile *profile=rtp_profile_clone_full(&av_profile);
OrtpEvQueue *q=ortp_ev_queue_new();
ms_init();
signal(SIGINT,stop_handler);
pt=rtp_profile_get_payload(profile,payload);
if (pt==NULL){
printf("Error: no payload defined with number %i.",payload);
exit(-1);
}
if (fmtp!=NULL) payload_type_set_send_fmtp(pt,fmtp);
if (bitrate>0) pt->normal_bitrate=bitrate;
if (pt->type!=PAYLOAD_VIDEO){
MSSndCardManager *manager=ms_snd_card_manager_get();
MSSndCard *capt= capture_card==NULL ? ms_snd_card_manager_get_default_capture_card(manager) :
ms_snd_card_manager_get_card(manager,capture_card);
MSSndCard *play= playback_card==NULL ? ms_snd_card_manager_get_default_playback_card(manager) :
ms_snd_card_manager_get_card(manager,playback_card);
audio=audio_stream_new(localport,ms_is_ipv6(remote_ip));
audio_stream_enable_automatic_gain_control(audio,agc);
audio_stream_enable_noise_gate(audio,use_ng);
audio_stream_set_echo_canceller_params(audio,ec_len_ms,ec_delay_ms,ec_framesize);
printf("Starting audio stream.\n");
audio_stream_start_full(audio,profile,remote_ip,remoteport,remoteport+1, payload, jitter,infile,outfile,
outfile==NULL ? play : NULL ,infile==NULL ? capt : NULL,infile!=NULL ? FALSE: ec);
if (audio) {
if (use_ng && ng_threshold!=-1)
ms_filter_call_method(audio->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_threshold);
session=audio->session;
}
}else{
#ifdef VIDEO_ENABLED
if (eq){
ms_fatal("Cannot put an audio equalizer in a video stream !");
exit(-1);
}
printf("Starting video stream.\n");
video=video_stream_new(localport, ms_is_ipv6(remote_ip));
video_stream_set_sent_video_size(video,vs);
video_stream_use_preview_video_window(video,two_windows);
video_stream_start(video,profile,
remote_ip,
remoteport,remoteport+1,
payload,
jitter,
ms_web_cam_manager_get_default_cam(ms_web_cam_manager_get()));
session=video->session;
#else
printf("Error: video support not compiled.\n");
#endif
}
if (eq || ec){ /*read from stdin interactive commands */
char commands[128];
commands[127]='\0';
ms_sleep(1); /* ensure following text be printed after ortp messages */
if (eq)
printf("\nPlease enter equalizer requests, such as 'eq active 1', 'eq active 0', 'eq 1200 0.1 200'\n");
if (ec)
printf("\nPlease enter echo canceller requests: ec reset; ec <delay ms> <tail_length ms'\n");
while(fgets(commands,sizeof(commands)-1,stdin)!=NULL){
int active,freq,freq_width;
int delay_ms, tail_ms;
float gain;
if (sscanf(commands,"eq active %i",&active)==1){
audio_stream_enable_equalizer(audio,active);
printf("OK\n");
}else if (sscanf(commands,"eq %i %f %i",&freq,&gain,&freq_width)==3){
audio_stream_equalizer_set_gain(audio,freq,gain,freq_width);
printf("OK\n");
}else if (sscanf(commands,"eq %i %f",&freq,&gain)==2){
audio_stream_equalizer_set_gain(audio,freq,gain,0);
printf("OK\n");
}else if (strstr(commands,"dump")){
int n=0,i;
float *t;
ms_filter_call_method(audio->equalizer,MS_EQUALIZER_GET_NUM_FREQUENCIES,&n);
t=(float*)alloca(sizeof(float)*n);
ms_filter_call_method(audio->equalizer,MS_EQUALIZER_DUMP_STATE,t);
for(i=0;i<n;++i){
if (fabs(t[i]-1)>0.01){
printf("%i:%f:0 ",(i*pt->clock_rate)/(2*n),t[i]);
}
}
printf("\nOK\n");
}else if (sscanf(commands,"ec reset %i",&active)==1){
//audio_stream_enable_equalizer(audio,active);
//printf("OK\n");
}else if (sscanf(commands,"ec active %i",&active)==1){
//audio_stream_enable_equalizer(audio,active);
//printf("OK\n");
}else if (sscanf(commands,"ec %i %i",&delay_ms,&tail_ms)==2){
audio_stream_set_echo_canceller_params(audio,tail_ms,delay_ms,128);
// revisit: workaround with old method call to force echo reset
delay_ms*=8;
ms_filter_call_method(audio->ec,MS_FILTER_SET_PLAYBACKDELAY,&delay_ms);
printf("OK\n");
}else if (strstr(commands,"quit")){
break;
}else printf("Cannot understand this.\n");
}
}else{ /* no interactive stuff - continuous debug output */
rtp_session_register_event_queue(session,q);
while(cond)
{
int n;
for(n=0;n<100;++n){
#ifdef WIN32
MSG msg;
Sleep(10);
while (PeekMessage(&msg, NULL, 0, 0,1)){
TranslateMessage(&msg);
DispatchMessage(&msg);
}
#else
struct timespec ts;
ts.tv_sec=0;
ts.tv_nsec=10000000;
nanosleep(&ts,NULL);
#endif
#if defined(VIDEO_ENABLED)
if (video) video_stream_iterate(video);
#endif
}
ortp_global_stats_display();
if (session){
printf("Bandwidth usage: download=%f kbits/sec, upload=%f kbits/sec\n",
rtp_session_compute_recv_bandwidth(session)*1e-3,
rtp_session_compute_send_bandwidth(session)*1e-3);
parse_events(q);
}
}
}
printf("stopping all...\n");
if (audio) audio_stream_stop(audio);
#ifdef VIDEO_ENABLED
if (video) video_stream_stop(video);
#endif
ortp_ev_queue_destroy(q);
rtp_profile_destroy(profile);
}
void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
LinphoneCore *lc=call->core;
LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
const char *tool="linphone-" LINPHONE_VERSION;
char *cname;
int used_pt=-1;
#ifdef VIDEO_ENABLED
const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpAvp,SalVideo);
#endif
bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
if(call->audiostream == NULL)
{
ms_fatal("start_media_stream() called without prior init !");
return;
}
call->current_params = call->params;
/* adjust rtp jitter compensation. It must be at least the latency of the sound card */
int jitt_comp=MAX(lc->sound_conf.latency,lc->rtp_conf.audio_jitt_comp);
if (call->media_start_time==0) call->media_start_time=time(NULL);
cname=linphone_address_as_string_uri_only(me);
{
const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpAvp,SalAudio);
if (stream && stream->dir!=SalStreamInactive && stream->port!=0){
MSSndCard *playcard=lc->sound_conf.lsd_card ?
lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
MSSndCard *captcard=lc->sound_conf.capt_sndcard;
const char *playfile=lc->play_file;
const char *recfile=lc->rec_file;
call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
bool_t use_ec,use_arc_audio=use_arc;
if (used_pt!=-1){
if (playcard==NULL) {
ms_warning("No card defined for playback !");
}
if (captcard==NULL) {
ms_warning("No card defined for capture !");
}
/*Replace soundcard filters by inactive file players or recorders
when placed in recvonly or sendonly mode*/
if (stream->port==0 || stream->dir==SalStreamRecvOnly){
captcard=NULL;
playfile=NULL;
}else if (stream->dir==SalStreamSendOnly){
playcard=NULL;
captcard=NULL;
recfile=NULL;
/*And we will eventually play "playfile" if set by the user*/
/*playfile=NULL;*/
}
if (send_ringbacktone){
captcard=NULL;
playfile=NULL;/* it is setup later*/
}
/*if playfile are supplied don't use soundcards*/
if (lc->use_files) {
captcard=NULL;
playcard=NULL;
}
use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
#if defined(VIDEO_ENABLED)
if (vstream && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
/*when video is used, do not make adaptive rate control on audio, it is stupid.*/
use_arc_audio=FALSE;
#if defined(ANDROID)
/*On android we have to disable the echo canceller to preserve CPU for video codecs */
use_ec=FALSE;
#endif
}
#endif
audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc_audio);
audio_stream_start_full(
call->audiostream,
call->audio_profile,
stream->addr[0]!='\0' ? stream->addr : call->resultdesc->addr,
stream->port,
stream->port+1,
used_pt,
jitt_comp,
playfile,
recfile,
playcard,
captcard,
use_ec
);
post_configure_audio_streams(call);
if (all_inputs_muted && !send_ringbacktone){
audio_stream_set_mic_gain(call->audiostream,0);
}
if (stream->dir==SalStreamSendOnly && playfile!=NULL){
int pause_time=500;
ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
}
if (send_ringbacktone){
setup_ring_player(lc,call);
}
audio_stream_set_rtcp_information(call->audiostream, cname, tool);
}else ms_warning("No audio stream accepted ?");
}
}
#ifdef VIDEO_ENABLED
{
used_pt=-1;
/* shutdown preview */
if (lc->previewstream!=NULL) {
video_preview_stop(lc->previewstream);
lc->previewstream=NULL;
}
call->current_params.has_video=FALSE;
if (vstream && vstream->dir!=SalStreamInactive && vstream->port!=0) {
const char *addr=vstream->addr[0]!='\0' ? vstream->addr : call->resultdesc->addr;
call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
if (used_pt!=-1){
VideoStreamDir dir=VideoStreamSendRecv;
MSWebCam *cam=lc->video_conf.device;
bool_t is_inactive=FALSE;
call->current_params.has_video=TRUE;
video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
if (lc->video_window_id!=0)
video_stream_set_native_window_id(call->videostream,lc->video_window_id);
if (lc->preview_window_id!=0)
video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
cam=get_nowebcam_device();
dir=VideoStreamSendOnly;
}else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
dir=VideoStreamRecvOnly;
}else if (vstream->dir==SalStreamSendRecv){
if (lc->video_conf.display && lc->video_conf.capture)
dir=VideoStreamSendRecv;
else if (lc->video_conf.display)
dir=VideoStreamRecvOnly;
else
dir=VideoStreamSendOnly;
}else{
ms_warning("video stream is inactive.");
/*either inactive or incompatible with local capabilities*/
is_inactive=TRUE;
}
if (call->camera_active==FALSE || all_inputs_muted){
cam=get_nowebcam_device();
}
if (!is_inactive){
video_stream_set_direction (call->videostream, dir);
video_stream_start(call->videostream,
call->video_profile, addr, vstream->port,
vstream->port+1,
used_pt, jitt_comp, cam);
video_stream_set_rtcp_information(call->videostream, cname,tool);
}
}else ms_warning("No video stream accepted.");
}else{
ms_warning("No valid video stream defined.");
}
}
#endif
call->all_muted=all_inputs_muted;
call->playing_ringbacktone=send_ringbacktone;
call->up_bw=linphone_core_get_upload_bandwidth(lc);
if (ortp_zrtp_available()) {
OrtpZrtpParams params;
params.zid=get_hexa_zrtp_identifier(lc);
params.zid_file=lc->zrtp_secrets_cache;
audio_stream_enable_zrtp(call->audiostream,¶ms);
}
goto end;
end:
ms_free(cname);
linphone_address_destroy(me);
}