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Logging: improve log formatting of statistics to align correctly

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Gautier Pelloux-Prayer 2015-03-09 11:56:20 +01:00
parent a69ea2cace
commit 1852c2ae0c
2 changed files with 29 additions and 29 deletions
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56
coreapi/linphonecall.c
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@ -240,7 +240,7 @@ static int get_max_codec_sample_rate(const MSList *codecs){
for(it=codecs;it!=NULL;it=it->next){
PayloadType *pt=(PayloadType*)it->data;
int sample_rate;
if( strcasecmp("G722",pt->mime_type) == 0 ){
/* G722 spec says 8000 but the codec actually requires 16000 */
sample_rate = 16000;
@ -283,7 +283,7 @@ static void linphone_core_assign_payload_type_numbers(LinphoneCore *lc, MSList *
for (elem=codecs; elem!=NULL; elem=elem->next){
PayloadType *pt=(PayloadType*)elem->data;
int number=payload_type_get_number(pt);
/*check if number is duplicated: it could be the case if the remote forced us to use a mapping with a previous offer*/
if (number!=-1 && !(pt->flags & PAYLOAD_TYPE_FROZEN_NUMBER)){
if (!is_payload_type_number_available(codecs, number, pt)){
@ -291,7 +291,7 @@ static void linphone_core_assign_payload_type_numbers(LinphoneCore *lc, MSList *
number=-1; /*need to be re-assigned*/
}
}
if (number==-1){
while(dyn_number<127){
if (is_payload_type_number_available(codecs, dyn_number, NULL)){
@ -364,7 +364,7 @@ static MSList *make_codec_list(LinphoneCore *lc, CodecConstraints * hints, SalSt
for(it=codecs;it!=NULL;it=it->next){
PayloadType *pt=(PayloadType*)it->data;
int num;
if (!(pt->flags & PAYLOAD_TYPE_ENABLED))
continue;
if (hints->bandwidth_limit>0 && !linphone_core_is_payload_type_usable_for_bandwidth(lc,pt,hints->bandwidth_limit)){
@ -376,14 +376,14 @@ static MSList *make_codec_list(LinphoneCore *lc, CodecConstraints * hints, SalSt
continue;
}
pt=payload_type_clone(pt);
/*look for a previously assigned number for this codec*/
num=find_payload_type_number(hints->previously_used, pt);
if (num!=-1){
payload_type_set_number(pt,num);
payload_type_set_flag(pt, PAYLOAD_TYPE_FROZEN_NUMBER);
}
l=ms_list_append(l, pt);
nb++;
if ((hints->max_codecs > 0) && (nb >= hints->max_codecs)) break;
@ -568,8 +568,8 @@ static const char *linphone_call_get_bind_ip_for_stream(LinphoneCall *call, int
static const char *linphone_call_get_public_ip_for_stream(LinphoneCall *call, int stream_index){
const char *public_ip=call->localip;
if (stream_index<2 && call->media_ports[stream_index].multicast_ip[0]!='\0')
if (stream_index<2 && call->media_ports[stream_index].multicast_ip[0]!='\0')
public_ip=call->media_ports[stream_index].multicast_ip;
return public_ip;
}
@ -584,7 +584,7 @@ void linphone_call_make_local_media_description(LinphoneCore *lc, LinphoneCall *
LinphoneAddress *addr;
const char *subject;
CodecConstraints codec_hints={0};
/*multicast is only set in case of outgoing call*/
if (call->dir == LinphoneCallOutgoing && linphone_core_audio_multicast_enabled(lc)) {
md->streams[0].ttl=linphone_core_get_audio_multicast_ttl(lc);
@ -803,7 +803,7 @@ static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from,
linphone_core_get_video_port_range(call->core, &min_port, &max_port);
port_config_set(call,1,min_port,max_port);
if (call->dir==LinphoneCallOutgoing){
if ( linphone_core_audio_multicast_enabled(call->core)){
strncpy(call->media_ports[0].multicast_ip,
@ -817,7 +817,7 @@ static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from,
linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
}
void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
@ -1295,7 +1295,7 @@ void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const
default:
break;
}
if(cstate!=LinphoneCallStreamsRunning) {
if (call->dtmfs_timer!=NULL){
/*cancelling DTMF sequence, if any*/
@ -1873,7 +1873,7 @@ void linphone_call_init_audio_stream(LinphoneCall *call){
if (call->audiostream != NULL) return;
if (call->sessions[0].rtp_session==NULL){
call->audiostream=audiostream=audio_stream_new2(linphone_call_get_bind_ip_for_stream(call,0),
call->media_ports[0].mcast_rtp_port ? call->media_ports[0].mcast_rtp_port : call->media_ports[0].rtp_port,
call->media_ports[0].mcast_rtp_port ? call->media_ports[0].mcast_rtp_port : call->media_ports[0].rtp_port,
call->media_ports[0].mcast_rtcp_port ? call->media_ports[0].mcast_rtcp_port : call->media_ports[0].rtcp_port);
linphone_call_join_multicast_group(call, 0, &audiostream->ms);
rtp_session_enable_network_simulation(call->audiostream->ms.sessions.rtp_session, &lc->net_conf.netsim_params);
@ -2430,7 +2430,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, b
playfile=lc->play_file;
recfile=lc->rec_file;
call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
if (used_pt!=-1){
call->current_params->audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
if (playcard==NULL) {
@ -2501,7 +2501,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, bool_t muted, b
configure_rtp_session_for_rtcp_xr(lc, call, SalAudio);
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,
@ -3301,7 +3301,7 @@ void linphone_call_stop_recording(LinphoneCall *call){
static void report_bandwidth(LinphoneCall *call, MediaStream *as, MediaStream *vs){
bool_t as_active = as ? (media_stream_get_state(as) == MSStreamStarted) : FALSE;
bool_t vs_active = vs ? (media_stream_get_state(vs) == MSStreamStarted) : FALSE;
call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth=(as_active) ? (media_stream_get_down_bw(as)*1e-3) : 0;
call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth=(as_active) ? (media_stream_get_up_bw(as)*1e-3) : 0;
call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth=(vs_active) ? (media_stream_get_down_bw(vs)*1e-3) : 0;
@ -3311,18 +3311,18 @@ static void report_bandwidth(LinphoneCall *call, MediaStream *as, MediaStream *v
call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_download_bandwidth=(vs_active) ? (media_stream_get_rtcp_down_bw(vs)*1e-3) : 0;
call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_upload_bandwidth=(vs_active) ? (media_stream_get_rtcp_up_bw(vs)*1e-3) : 0;
ms_message("Bandwidth usage for call [%p]: RTP audio=[d=%.1f,u=%.1f], video=[d=%.1f,u=%.1f] kbit/sec",
call,
call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth,
call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth ,
call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth,
call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth
);
ms_message(" RTCP audio=[d=%.1f,u=%.1f], video=[d=%.1f,u=%.1f] kbit/sec",
call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_download_bandwidth,
call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_upload_bandwidth ,
call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_download_bandwidth,
call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_upload_bandwidth
ms_message( "Bandwidth usage for call [%p]:\n"
"\tRTP audio=[d=%5.1f,u=%5.1f], video=[d=%5.1f,u=%5.1f] kbits/sec\n"
"\tRTCP audio=[d=%5.1f,u=%5.1f], video=[d=%5.1f,u=%5.1f] kbits/sec",
call,
call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth,
call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth,
call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth,
call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth,
call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_download_bandwidth,
call->stats[LINPHONE_CALL_STATS_AUDIO].rtcp_upload_bandwidth,
call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_download_bandwidth,
call->stats[LINPHONE_CALL_STATS_VIDEO].rtcp_upload_bandwidth
);
}

2
mediastreamer2

@ -1 +1 @@
Subproject commit 4c95c0b22b53d594d8a531bc343b39ee6ec01aa5
Subproject commit 04d3a58d3f60dedcfe944601d0db4eef07f09cdc