Commit 98c4a9bb authored by Simon Morlat's avatar Simon Morlat
Browse files

wip - conferencing

parent 1caf08c5
......@@ -36,7 +36,8 @@ liblinphone_la_SOURCES=\
sipsetup.c sipsetup.h \
siplogin.c \
lsd.c linphonecore_utils.h \
ec-calibrator.c
ec-calibrator.c \
conference.c
liblinphone_la_LDFLAGS= -version-info $(LIBLINPHONE_SO_VERSION) -no-undefined
......
/***************************************************************************
* conference.c
*
* Mon Sep 12, 2011
* Copyright 2011 Belledonne Communications
* Author: Simon Morlat
* Email simon dot morlat at linphone dot org
****************************************************************************/
/*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#include "private.h"
static void conference_check_init(LinphoneConference *ctx){
if (ctx->conf==NULL){
ctx->conf=ms_audio_conference_new();
}
}
static void conference_check_uninit(LinphoneConference *ctx){
if (ctx->conf){
if (ctx->conf->nmembers==0){
ms_audio_conference_destroy(ctx->conf);
ctx->conf=NULL;
}
}
}
void linphone_call_add_to_conf(LinphoneCall *call){
LinphoneCore *lc=call->core;
LinphoneConference *conf=&lc->conf_ctx;
MSAudioEndpoint *ep;
ep=ms_audio_endpoint_get_from_stream(call->audiostream,TRUE);
ms_audio_conference_add_member(conf->conf,ep);
call->endpoint=ep;
}
void linphone_call_remove_from_conf(LinphoneCall *call){
LinphoneCore *lc=call->core;
LinphoneConference *conf=&lc->conf_ctx;
ms_audio_conference_remove_member(conf->conf,call->endpoint);
ms_audio_endpoint_release_from_stream(call->endpoint);
call->endpoint=NULL;
conference_check_uninit(conf);
}
int linphone_core_add_to_conference(LinphoneCore *lc, LinphoneCall *call){
LinphoneCallParams params;
if (call->current_params.in_conference){
ms_error("Already in conference");
return -1;
}
conference_check_init(&lc->conf_ctx);
call->params.in_conference=TRUE;
call->params.has_video=FALSE;
params=call->params;
if (call->state==LinphoneCallPaused)
linphone_core_resume_call(lc,call);
else if (call->state==LinphoneCallStreamsRunning){
/*this will trigger a reINVITE that will later redraw the streams */
linphone_core_update_call(lc,call,&params);
}else{
ms_error("Call is in state %s, it cannot be added to the conference.",linphone_call_state_to_string(call->state));
return -1;
}
return 0;
}
int linphone_core_remove_from_conference(LinphoneCore *lc, LinphoneCall *call){
if (!call->current_params.in_conference){
if (call->params.in_conference){
ms_warning("Not (yet) in conference, be patient");
return -1;
}else{
ms_error("Not in a conference.");
return -1;
}
}
call->params.in_conference=FALSE;
return linphone_core_pause_call(lc,call);
}
int linphone_core_pause_conference(LinphoneCore *lc){
return 0;
}
int linphone_core_resume_conference(LinphoneCore *lc){
return 0;
}
......@@ -834,10 +834,7 @@ static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
}
}
static void post_configure_audio_streams(LinphoneCall*call){
AudioStream *st=call->audiostream;
LinphoneCore *lc=call->core;
static void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
float thres = 0;
float recv_gain;
......@@ -845,7 +842,7 @@ static void post_configure_audio_streams(LinphoneCall*call){
float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
if (!call->audio_muted)
if (!muted)
audio_stream_set_mic_gain(st,mic_gain);
else
audio_stream_set_mic_gain(st,0);
......@@ -884,6 +881,12 @@ static void post_configure_audio_streams(LinphoneCall*call){
ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
}
parametrize_equalizer(lc,st);
}
static void post_configure_audio_streams(LinphoneCall*call){
AudioStream *st=call->audiostream;
LinphoneCore *lc=call->core;
_post_configure_audio_stream(st,lc,call->audio_muted);
if (lc->vtable.dtmf_received!=NULL){
/* replace by our default action*/
audio_stream_play_received_dtmfs(call->audiostream,FALSE);
......@@ -953,18 +956,167 @@ static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
}
#define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
LinphoneCore *lc=call->core;
int jitt_comp=lc->rtp_conf.audio_jitt_comp;
int used_pt=-1;
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;
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;
}
if (call->params.in_conference){
/* first create the graph without soundcard resources*/
captcard=playcard=NULL;
}
use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
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 (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, LINPHONE_RTCP_SDES_TOOL);
if (call->params.in_conference){
/*transform the graph to connect it to the conference filter */
linphone_call_add_to_conf(call);
}
}else ms_warning("No audio stream accepted ?");
}
}
static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
#ifdef VIDEO_ENABLED
LinphoneCore *lc=call->core;
int used_pt=-1;
const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpAvp,SalVideo);
/* 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, lc->rtp_conf.audio_jitt_comp, cam);
video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
}
}else ms_warning("No video stream accepted.");
}else{
ms_warning("No valid video stream defined.");
}
#endif
}
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;
bool_t use_arc;
#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)
{
......@@ -972,155 +1124,18 @@ void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_mut
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.");
}
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=FALSE;
}
#endif
linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
linphone_call_start_video_stream(call,cname,all_inputs_muted);
call->all_muted=all_inputs_muted;
call->playing_ringbacktone=send_ringbacktone;
call->up_bw=linphone_core_get_upload_bandwidth(lc);
......@@ -1158,6 +1173,9 @@ void linphone_call_stop_media_streams(LinphoneCall *call){
}
}
linphone_call_log_fill_stats (call->log,call->audiostream);
if (call->endpoint){
linphone_call_remove_from_conf(call);
}
audio_stream_stop(call->audiostream);
call->audiostream=NULL;
}
......
......@@ -2219,13 +2219,20 @@ bool_t linphone_core_inc_invite_pending(LinphoneCore*lc){
int linphone_core_update_call(LinphoneCore *lc, LinphoneCall *call, const LinphoneCallParams *params){
int err=0;
if (params!=NULL){
const char *subject;
call->params=*params;
update_local_media_description(lc,call,&call->localdesc);
call->camera_active=params->has_video;
if (params->in_conference){
subject="Conference";
}else{
subject="Media change";
}
if (lc->vtable.display_status)
lc->vtable.display_status(lc,_("Modifying call parameters..."));
sal_call_set_local_media_description (call->op,call->localdesc);
err=sal_call_update(call->op,"Media parameters update");
err=sal_call_update(call->op,subject);
}else{
#ifdef VIDEO_ENABLED
if (call->videostream!=NULL){
......@@ -2486,8 +2493,7 @@ int linphone_core_pause_call(LinphoneCore *lc, LinphoneCall *the_call)
ms_error("No reason to pause this call, it is already paused or inactive.");
return -1;
}
if (sal_call_update(call->op,subject) != 0)
{
if (sal_call_update(call->op,subject) != 0){
if (lc->vtable.display_warning)
lc->vtable.display_warning(lc,_("Could not pause the call"));
}
......@@ -2524,6 +2530,7 @@ int linphone_core_resume_call(LinphoneCore *lc, LinphoneCall *the_call)
{
char temp[255]={0};
LinphoneCall *call = the_call;
const char *subject="Call resuming";
if(call->state!=LinphoneCallPaused ){
ms_warning("we cannot resume a call that has not been established and paused before");
......@@ -2537,7 +2544,8 @@ int linphone_core_resume_call(LinphoneCore *lc, LinphoneCall *the_call)
}
ms_message("Resuming call %p",call);
sal_media_description_set_dir(call->localdesc,SalStreamSendRecv);
if(sal_call_update(call->op,"Call resuming") != 0){
if (call->params.in_conference) subject="Resuming conference";
if(sal_call_update(call->op,subject) != 0){
return -1;
}
linphone_call_set_state (call,LinphoneCallResuming,"Resuming");
......
......@@ -1023,7 +1023,10 @@ bool_t linphone_call_are_all_streams_encrypted(LinphoneCall *call);
const char* linphone_call_get_authentication_token(LinphoneCall *call);
bool_t linphone_call_get_authentication_token_verified(LinphoneCall *call);
int linphone_core_add_to_conference(LinphoneCore *lc, LinphoneCall *call);
int linphone_core_remove_from_conference(LinphoneCore *lc, LinphoneCall *call);
int linphone_core_pause_conference(LinphoneCore *lc);
int linphone_core_resume_conference(LinphoneCore *lc);
#ifdef __cplusplus
......
......@@ -33,6 +33,7 @@
#include "config.h"
#endif
#include "mediastreamer2/mediastream.h"
#include "mediastreamer2/msconference.h"
#ifndef LIBLINPHONE_VERSION
#define LIBLINPHONE_VERSION LINPHONE_VERSION
......@@ -62,7 +63,8 @@ struct _LinphoneCallParams{
int audio_bw; /* bandwidth limit for audio stream */
bool_t has_video;
bool_t real_early_media; /*send real media even during early media (for outgoing calls)*/
bool_t pad[2];
bool_t in_conference; /*in conference mode */
bool_t pad;
};
struct _LinphoneCall
......@@ -87,6 +89,7 @@ struct _LinphoneCall
int video_port;
struct _AudioStream *audiostream; /**/
struct _VideoStream *videostream;
MSAudioEndpoint *endpoint; /*used for conferencing*/
char *refer_to;
LinphoneCallParams params;
LinphoneCallParams current_params;
......@@ -99,12 +102,13 @@ struct _LinphoneCall
bool_t all_muted; /*this flag is set during early medias*/
bool_t playing_ringbacktone;
bool_t owns_call_log;
bool_t pad;
OrtpEvQueue *audiostream_app_evq;
bool_t audiostream_encrypted;
char *auth_token;
bool_t auth_token_verified;
OrtpEvQueue *videostream_app_evq;
bool_t videostream_encrypted;
bool_t audiostream_encrypted;
bool_t auth_token_verified;
};
......@@ -388,6 +392,13 @@ typedef struct autoreplier_config
const char *message; /* the path of the file to be played */
}autoreplier_config_t;
struct _LinphoneConference{
MSAudioConference *conf;
AudioStream *local_participant;
MSAudioEndpoint *local_endpoint;
};
typedef struct _LinphoneConference LinphoneConference;
struct _LinphoneCore
{
......@@ -439,6 +450,8 @@ struct _LinphoneCore