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Allow linphone to be configured to use RTP sessions for input/output of audio and video streams.

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This commit is contained in:
Ghislain MARY 2015-06-18 14:33:11 +02:00
parent f336bed4e6
commit 98f2508be5
6 changed files with 123 additions and 37 deletions
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127
coreapi/linphonecall.c
View file

@ -2513,6 +2513,30 @@ void static set_dtls_fingerprint_on_all_streams(LinphoneCall *call) {
return;
}
static RtpSession * create_audio_rtp_io_session(LinphoneCall *call) {
PayloadType *pt;
LinphoneCore *lc = call->core;
const char *local_ip = lp_config_get_string(lc->config, "sound", "rtp_local_addr", "127.0.0.1");
const char *remote_ip = lp_config_get_string(lc->config, "sound", "rtp_remote_addr", "127.0.0.1");
int local_port = lp_config_get_int(lc->config, "sound", "rtp_local_port", 17076);
int remote_port = lp_config_get_int(lc->config, "sound", "rtp_remote_port", 17078);
int ptnum = lp_config_get_int(lc->config, "sound", "rtp_ptnum", 0);
const char *rtpmap = lp_config_get_string(lc->config, "sound", "rtp_map", "pcmu/8000/1");
RtpSession *rtp_session = NULL;
pt = rtp_profile_get_payload_from_rtpmap(call->audio_profile, rtpmap);
if (pt != NULL) {
call->rtp_io_audio_profile = rtp_profile_new("RTP IO audio profile");
rtp_profile_set_payload(call->rtp_io_audio_profile, ptnum, payload_type_clone(pt));
rtp_session = ms_create_duplex_rtp_session(local_ip, local_port, -1);
rtp_session_set_profile(rtp_session, call->rtp_io_audio_profile);
rtp_session_set_remote_addr_and_port(rtp_session, remote_ip, remote_port, -1);
rtp_session_enable_rtcp(rtp_session, FALSE);
rtp_session_set_payload_type(rtp_session, ptnum);
rtp_session_set_jitter_compensation(rtp_session, linphone_core_get_audio_jittcomp(lc));
}
return rtp_session;
}
static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
LinphoneCore *lc=call->core;
LpConfig* conf;
@ -2527,6 +2551,8 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, b
const char *recfile;
const SalStreamDescription *local_st_desc;
int crypto_idx;
AudioStreamIO io = { 0 };
bool_t use_rtp_io = lp_config_get_int(lc->config, "sound", "rtp_io", FALSE);
snprintf(rtcp_tool,sizeof(rtcp_tool)-1,"%s-%s",linphone_core_get_user_agent_name(),linphone_core_get_user_agent_version());
@ -2542,6 +2568,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, b
call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
if (used_pt!=-1){
bool_t ok = TRUE;
call->current_params->audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
if (playcard==NULL) {
ms_warning("No card defined for playback !");
@ -2573,7 +2600,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, b
}
}
/*if playfile are supplied don't use soundcards*/
if (lc->use_files) {
if (lc->use_files || use_rtp_io) {
captcard=NULL;
playcard=NULL;
}
@ -2612,22 +2639,32 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, b
if (is_multicast)
rtp_session_set_multicast_ttl(call->audiostream->ms.sessions.rtp_session,stream->ttl);
audio_stream_start_full(
call->audiostream,
call->audio_profile,
rtp_addr,
stream->rtp_port,
stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
(linphone_core_rtcp_enabled(lc) && !is_multicast) ? (stream->rtcp_port ? stream->rtcp_port : stream->rtp_port+1) : 0,
used_pt,
linphone_core_get_audio_jittcomp(lc),
playfile,
recfile,
playcard,
captcard,
use_ec
if (lc->use_files) {
io.input_file = playfile;
io.output_file = recfile;
} else if (use_rtp_io) {
io.rtp_session = create_audio_rtp_io_session(call);
if (io.rtp_session == NULL) {
ok = FALSE;
}
} else {
io.playback_card = playcard;
io.capture_card = captcard;
}
if (ok == TRUE) {
audio_stream_start_from_io(call->audiostream,
call->audio_profile,
rtp_addr,
stream->rtp_port,
stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
(linphone_core_rtcp_enabled(lc) && !is_multicast) ? (stream->rtcp_port ? stream->rtcp_port : stream->rtp_port+1) : 0,
used_pt,
linphone_core_get_audio_jittcomp(lc),
use_ec,
&io
);
post_configure_audio_streams(call, muted && !send_ringbacktone);
post_configure_audio_streams(call, muted && !send_ringbacktone);
}
media_stream_session_encryption_mandatory_enable(&call->audiostream->ms.sessions,linphone_core_is_media_encryption_mandatory(call->core));
@ -2650,6 +2687,29 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, b
}
}
static RtpSession * create_video_rtp_io_session(LinphoneCall *call) {
PayloadType *pt;
LinphoneCore *lc = call->core;
const char *local_ip = lp_config_get_string(lc->config, "video", "rtp_local_addr", "127.0.0.1");
const char *remote_ip = lp_config_get_string(lc->config, "video", "rtp_remote_addr", "127.0.0.1");
int local_port = lp_config_get_int(lc->config, "video", "rtp_local_port", 19076);
int remote_port = lp_config_get_int(lc->config, "video", "rtp_remote_port", 19078);
int ptnum = lp_config_get_int(lc->config, "video", "rtp_ptnum", 0);
const char *rtpmap = lp_config_get_string(lc->config, "video", "rtp_map", "vp8/90000/1");
RtpSession *rtp_session = NULL;
pt = rtp_profile_get_payload_from_rtpmap(call->video_profile, rtpmap);
if (pt != NULL) {
call->rtp_io_video_profile = rtp_profile_new("RTP IO video profile");
rtp_profile_set_payload(call->rtp_io_video_profile, ptnum, payload_type_clone(pt));
rtp_session = ms_create_duplex_rtp_session(local_ip, local_port, -1);
rtp_session_set_profile(rtp_session, call->rtp_io_video_profile);
rtp_session_set_remote_addr_and_port(rtp_session, remote_ip, remote_port, -1);
rtp_session_enable_rtcp(rtp_session, FALSE);
rtp_session_set_payload_type(rtp_session, ptnum);
}
return rtp_session;
}
static void linphone_call_start_video_stream(LinphoneCall *call, bool_t all_inputs_muted){
#ifdef VIDEO_ENABLED
LinphoneCore *lc=call->core;
@ -2657,7 +2717,8 @@ static void linphone_call_start_video_stream(LinphoneCall *call, bool_t all_inpu
const SalStreamDescription *vstream;
MSFilter* source = NULL;
bool_t reused_preview = FALSE;
bool_t use_rtp_io = lp_config_get_int(lc->config, "video", "rtp_io", FALSE);
VideoStreamIO io = { 0 };
/* shutdown preview */
if (lc->previewstream!=NULL) {
@ -2746,7 +2807,7 @@ static void linphone_call_start_video_stream(LinphoneCall *call, bool_t all_inpu
rtp_session_set_multicast_ttl(call->videostream->ms.sessions.rtp_session,vstream->ttl);
video_stream_use_video_preset(call->videostream, lp_config_get_string(lc->config, "video", "preset", NULL));
if( lc->video_conf.reuse_preview_source && source ){
if (lc->video_conf.reuse_preview_source && source) {
ms_message("video_stream_start_with_source kept: %p", source);
video_stream_start_with_source(call->videostream,
call->video_profile, rtp_addr, vstream->rtp_port,
@ -2755,11 +2816,23 @@ static void linphone_call_start_video_stream(LinphoneCall *call, bool_t all_inpu
used_pt, linphone_core_get_video_jittcomp(lc), cam, source);
reused_preview = TRUE;
} else {
video_stream_start(call->videostream,
call->video_profile, rtp_addr, vstream->rtp_port,
rtcp_addr,
(linphone_core_rtcp_enabled(lc) && !is_multicast) ? (vstream->rtcp_port ? vstream->rtcp_port : vstream->rtp_port+1) : 0,
used_pt, linphone_core_get_video_jittcomp(lc), cam);
bool_t ok = TRUE;
if (use_rtp_io) {
io.rtp_session = create_video_rtp_io_session(call);
if (io.rtp_session == NULL) {
ok = FALSE;
ms_warning("Cannot create video RTP IO session");
}
} else {
io.cam = cam;
}
if (ok) {
video_stream_start_from_io(call->videostream,
call->video_profile, rtp_addr, vstream->rtp_port,
rtcp_addr,
(linphone_core_rtcp_enabled(lc) && !is_multicast) ? (vstream->rtcp_port ? vstream->rtcp_port : vstream->rtp_port+1) : 0,
used_pt, linphone_core_get_video_jittcomp(lc), &io);
}
}
media_stream_session_encryption_mandatory_enable(&call->videostream->ms.sessions,linphone_core_is_media_encryption_mandatory(call->core));
}
@ -3056,6 +3129,14 @@ void linphone_call_stop_media_streams(LinphoneCall *call){
call->video_profile=NULL;
unset_rtp_profile(call,1);
}
if (call->rtp_io_audio_profile) {
rtp_profile_destroy(call->rtp_io_audio_profile);
call->rtp_io_audio_profile = NULL;
}
if (call->rtp_io_video_profile) {
rtp_profile_destroy(call->rtp_io_video_profile);
call->rtp_io_video_profile = NULL;
}
}

3
coreapi/linphonecore.c
View file

@ -6527,6 +6527,7 @@ void linphone_core_soundcard_hint_check( LinphoneCore* lc){
MSList* the_calls = lc->calls;
LinphoneCall* call = NULL;
bool_t dont_need_sound = TRUE;
bool_t use_rtp_io = lp_config_get_int(lc->config, "sound", "rtp_input", FALSE);
/* check if the remaining calls are paused */
while( the_calls ){
@ -6539,7 +6540,7 @@ void linphone_core_soundcard_hint_check( LinphoneCore* lc){
}
/* if no more calls or all calls are paused, we can free the soundcard */
if ( (lc->calls==NULL || dont_need_sound) && !lc->use_files){
if ( (lc->calls==NULL || dont_need_sound) && !lc->use_files && !use_rtp_io){
ms_message("Notifying soundcard that we don't need it anymore for calls.");
notify_soundcard_usage(lc,FALSE);
}

2
coreapi/private.h
View file

@ -222,6 +222,8 @@ struct _LinphoneCall{
SalMediaDescription *resultdesc;
struct _RtpProfile *audio_profile;
struct _RtpProfile *video_profile;
struct _RtpProfile *rtp_io_audio_profile;
struct _RtpProfile *rtp_io_video_profile;
struct _LinphoneCallLog *log;
LinphoneAddress *me; /*Either from or to based on call dir*/
SalOp *op;

24
gtk/videowindow.c
View file

@ -292,17 +292,19 @@ void linphone_gtk_in_call_show_video(LinphoneCall *call){
GtkWidget *video_window=(GtkWidget*)g_object_get_data(G_OBJECT(callview),"video_window");
const LinphoneCallParams *params=linphone_call_get_current_params(call);
LinphoneCore *lc=linphone_gtk_get_core();
if (linphone_call_get_state(call)!=LinphoneCallPaused && params && linphone_call_params_video_enabled(params)){
if (video_window==NULL){
video_window=create_video_window(call);
g_object_set_data(G_OBJECT(callview),"video_window",video_window);
}
linphone_core_set_native_video_window_id(lc,get_native_handle(gtk_widget_get_window(video_window)));
}else{
if (video_window){
gtk_widget_destroy(video_window);
g_object_set_data(G_OBJECT(callview),"video_window",NULL);
if (((bool_t)lp_config_get_int(linphone_core_get_config(lc), "video", "rtp_io", FALSE)) == FALSE) {
if (linphone_call_get_state(call)!=LinphoneCallPaused && params && linphone_call_params_video_enabled(params)){
if (video_window==NULL){
video_window=create_video_window(call);
g_object_set_data(G_OBJECT(callview),"video_window",video_window);
}
linphone_core_set_native_video_window_id(lc,get_native_handle(gtk_widget_get_window(video_window)));
}else{
if (video_window){
gtk_widget_destroy(video_window);
g_object_set_data(G_OBJECT(callview),"video_window",NULL);
}
}
}
}

2
mediastreamer2

@ -1 +1 @@
Subproject commit 85f0d427898c83fe1f92d735f8d53198fc77d145
Subproject commit d348ca973af095425510e184b6800413a0dd0c14

2
oRTP

@ -1 +1 @@
Subproject commit 7170e5c6242b305114c681940ec78260847122ff
Subproject commit 75baaa5353f202172428c106e86202242e264a0f