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fix(coreapi/linphonecall): linphone_call_get_duration returns call->log->duration value at end call

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This commit is contained in:
Ronan Abhamon 2017-01-25 15:24:35 +01:00
parent a4109f7c4f
commit 9b6643b651
1 changed files with 22 additions and 8 deletions
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30
coreapi/linphonecall.c
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@ -2092,9 +2092,23 @@ bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
return call->refer_pending;
}
static int _linphone_call_get_duration (const LinphoneCall *call) {
return (int)(ms_time(NULL) - call->log->connected_date_time);
}
int linphone_call_get_duration(const LinphoneCall *call){
if (call->log->connected_date_time==0) return 0;
return (int)(ms_time(NULL) - call->log->connected_date_time);
switch (call->state) {
case LinphoneCallEnd:
case LinphoneCallError:
case LinphoneCallReleased:
return call->log->duration;
default: break;
}
return _linphone_call_get_duration(call);
}
LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
@ -2436,7 +2450,7 @@ static OrtpJitterBufferAlgorithm name_to_jb_algo(const char *value){
static void apply_jitter_buffer_params(LinphoneCore *lc, RtpSession *session, LinphoneStreamType type){
JBParameters params;
rtp_session_get_jitter_buffer_params(session, &params);
params.min_size = lp_config_get_int(lc->config, "rtp", "jitter_buffer_min_size", 40);
params.max_size = lp_config_get_int(lc->config, "rtp", "jitter_buffer_max_size", 500);
@ -2446,7 +2460,7 @@ static void apply_jitter_buffer_params(LinphoneCore *lc, RtpSession *session, Li
params.ramp_refresh_ms = lp_config_get_int(lc->config, "rtp", "jitter_buffer_ramp_refresh_period", 5000);
params.ramp_step_ms = lp_config_get_int(lc->config, "rtp", "jitter_buffer_ramp_step", 20);
params.ramp_threshold = lp_config_get_int(lc->config, "rtp", "jitter_buffer_ramp_threshold", 70);
switch (type){
case LinphoneStreamTypeAudio:
case LinphoneStreamTypeText: /*let's use the same params for text as for audio.*/
@ -3221,7 +3235,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, LinphoneCallSta
}
}
/*if playfile are supplied don't use soundcards*/
if (lc->use_files || (use_rtp_io && !use_rtp_io_enable_local_output)) {
if (lc->use_files || (use_rtp_io && !use_rtp_io_enable_local_output)) {
captcard=NULL;
playcard=NULL;
}
@ -3240,7 +3254,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, LinphoneCallSta
audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
media_stream_set_adaptive_bitrate_algorithm(&call->audiostream->ms,
ms_qos_analyzer_algorithm_from_string(linphone_core_get_adaptive_rate_algorithm(lc)));
rtp_session_enable_rtcp_mux(call->audiostream->ms.sessions.rtp_session, stream->rtcp_mux);
if (!call->params->in_conference && call->params->record_file){
audio_stream_mixed_record_open(call->audiostream,call->params->record_file);
@ -3267,7 +3281,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, LinphoneCallSta
if(use_rtp_io_enable_local_output){
io.input.type = MSResourceRtp;
io.input.session = create_audio_rtp_io_session(call);
if (playcard){
io.output.type = MSResourceSoundcard;
io.output.soundcard = playcard;
@ -3280,7 +3294,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, LinphoneCallSta
io.input.type = io.output.type = MSResourceRtp;
io.input.session = io.output.session = create_audio_rtp_io_session(call);
}
if (io.input.session == NULL) {
ok = FALSE;
}
@ -4569,7 +4583,7 @@ void linphone_call_log_completed(LinphoneCall *call){
LinphoneCore *lc=call->core;
bool_t call_logs_sqlite_db_found = FALSE;
call->log->duration=linphone_call_get_duration(call); /*store duration since connected*/
call->log->duration=_linphone_call_get_duration(call); /*store duration since connected*/
if (call->log->status==LinphoneCallMissed){
char *info;