Commit 7e06844a authored by Simon Morlat's avatar Simon Morlat

enhance liblinphone documentation

parent a2019a94
......@@ -922,11 +922,14 @@ bool_t linphone_core_tunnel_available(void){
}
/**
* Enable adaptive rate control (experimental feature, audio-only).
* Enable adaptive rate control.
*
* @ingroup media_parameters
*
* Adaptive rate control consists in using RTCP feedback provided information to dynamically
* control the output bitrate of the encoders, so that we can adapt to the network conditions and
* available bandwidth.
* control the output bitrate of the audio and video encoders, so that we can adapt to the network conditions and
* available bandwidth. Control of the audio encoder is done in case of audio-only call, and control of the video encoder is done for audio & video calls.
* Adaptive rate control feature is enabled by default.
**/
void linphone_core_enable_adaptive_rate_control(LinphoneCore *lc, bool_t enabled){
lp_config_set_int(lc->config,"net","adaptive_rate_control",(int)enabled);
......@@ -934,6 +937,8 @@ void linphone_core_enable_adaptive_rate_control(LinphoneCore *lc, bool_t enabled
/**
* Returns whether adaptive rate control is enabled.
*
* @ingroup media_parameters
*
* See linphone_core_enable_adaptive_rate_control().
**/
......@@ -1006,14 +1011,18 @@ int linphone_core_get_upload_bandwidth(const LinphoneCore *lc){
return lc->net_conf.upload_bw;
}
/**
* Set audio packetization time linphone expects to receive from peer
* Set audio packetization time linphone expects to receive from peer.
* A value of zero means that ptime is not specified.
* @ingroup media_parameters
*/
void linphone_core_set_download_ptime(LinphoneCore *lc, int ptime) {
lp_config_set_int(lc->config,"rtp","download_ptime",ptime);
}
/**
* Get audio packetization time linphone expects to receive from peer
* Get audio packetization time linphone expects to receive from peer.
* A value of zero means that ptime is not specified.
* @ingroup media_parameters
*/
int linphone_core_get_download_ptime(LinphoneCore *lc) {
return lp_config_get_int(lc->config,"rtp","download_ptime",0);
......@@ -1023,6 +1032,7 @@ int linphone_core_get_download_ptime(LinphoneCore *lc) {
* Set audio packetization time linphone will send (in absence of requirement from peer)
* A value of 0 stands for the current codec default packetization time.
*
* @ingroup media_parameters
**/
void linphone_core_set_upload_ptime(LinphoneCore *lc, int ptime){
lp_config_set_int(lc->config,"rtp","upload_ptime",ptime);
......@@ -1032,6 +1042,8 @@ void linphone_core_set_upload_ptime(LinphoneCore *lc, int ptime){
* Set audio packetization time linphone will send (in absence of requirement from peer)
* A value of 0 stands for the current codec default packetization time.
*
*
* @ingroup media_parameters
**/
int linphone_core_get_upload_ptime(LinphoneCore *lc){
return lp_config_get_int(lc->config,"rtp","upload_ptime",0);
......@@ -1265,6 +1277,7 @@ LinphoneCore *linphone_core_new(const LinphoneCoreVTable *vtable,
* structure holding the codec information.
* It is possible to make copy of the list with ms_list_copy() in order to modify it
* (such as the order of codecs).
* @ingroup media_parameters
**/
const MSList *linphone_core_get_audio_codecs(const LinphoneCore *lc)
{
......@@ -1278,6 +1291,7 @@ const MSList *linphone_core_get_audio_codecs(const LinphoneCore *lc)
* structure holding the codec information.
* It is possible to make copy of the list with ms_list_copy() in order to modify it
* (such as the order of codecs).
* @ingroup media_parameters
**/
const MSList *linphone_core_get_video_codecs(const LinphoneCore *lc)
{
......@@ -1567,6 +1581,7 @@ void linphone_core_set_audio_port(LinphoneCore *lc, int port)
/**
* Sets the UDP port range from which to randomly select the port used for audio streaming.
* @ingroup media_parameters
*/
void linphone_core_set_audio_port_range(LinphoneCore *lc, int min_port, int max_port)
{
......@@ -1585,6 +1600,7 @@ void linphone_core_set_video_port(LinphoneCore *lc, int port){
/**
* Sets the UDP port range from which to randomly select the port used for video streaming.
* @ingroup media_parameters
*/
void linphone_core_set_video_port_range(LinphoneCore *lc, int min_port, int max_port)
{
......@@ -2054,6 +2070,8 @@ void linphone_core_iterate(LinphoneCore *lc){
/**
* Interpret a call destination as supplied by the user, and returns a fully qualified
* LinphoneAddress.
*
* @ingroup call_control
*
* A sip address should look like DisplayName <sip:username@domain:port> .
* Basically this function performs the following tasks
......@@ -2498,6 +2516,7 @@ LinphoneCall * linphone_core_invite_address_with_params(LinphoneCore *lc, const
/**
* Performs a simple call transfer to the specified destination.
*
* @ingroup call_control
* The remote endpoint is expected to issue a new call to the specified destination.
* The current call remains active and thus can be later paused or terminated.
**/
......@@ -2528,6 +2547,8 @@ int linphone_core_transfer_call(LinphoneCore *lc, LinphoneCall *call, const char
* @param lc linphone core object
* @param call a running call you want to transfer
* @param dest a running call whose remote person will receive the transfer
*
* @ingroup call_control
*
* The transfered call is supposed to be in paused state, so that it is able to accept the transfer immediately.
* The destination call is a call previously established to introduce the transfered person.
......@@ -3024,6 +3045,9 @@ int linphone_core_terminate_call(LinphoneCore *lc, LinphoneCall *the_call)
/**
* Decline a pending incoming call, with a reason.
*
* @ingroup call_control
*
* @param lc the linphone core
* @param call the LinphoneCall, must be in the IncomingReceived state.
* @param reason the reason for rejecting the call: LinphoneReasonDeclined or LinphoneReasonBusy
......@@ -3143,6 +3167,7 @@ int linphone_core_pause_call(LinphoneCore *lc, LinphoneCall *call)
/**
* Pause all currently running calls.
* @ingroup call_control
**/
int linphone_core_pause_all_calls(LinphoneCore *lc){
const MSList *elem;
......@@ -3219,6 +3244,8 @@ static int remote_address_compare(LinphoneCall *call, const LinphoneAddress *rad
* @param lc
* @param remote_address
* @return the LinphoneCall of the call if found
*
* @ingroup call_control
*/
LinphoneCall *linphone_core_get_call_by_remote_address(LinphoneCore *lc, const char *remote_address){
LinphoneAddress *raddr=linphone_address_new(remote_address);
......@@ -3692,18 +3719,18 @@ const char *linphone_core_get_ring(const LinphoneCore *lc){
* @param path
* @param lc The LinphoneCore object
*
* @ingroup media_parameters
* @ingroup initializing
**/
void linphone_core_set_root_ca(LinphoneCore *lc,const char *path){
sal_set_root_ca(lc->sal, path);
}
/**
* Gets the path to a file or folder containing trusted root CAs (PEM format)
* Gets the path to a file or folder containing the trusted root CAs (PEM format)
*
* @param lc The LinphoneCore object
*
* @ingroup media_parameters
* @ingroup initializing
**/
const char *linphone_core_get_root_ca(LinphoneCore *lc){
return sal_get_root_ca(lc->sal);
......@@ -3711,6 +3738,8 @@ const char *linphone_core_get_root_ca(LinphoneCore *lc){
/**
* Specify whether the tls server certificate must be verified when connecting to a SIP/TLS server.
*
* @ingroup initializing
**/
void linphone_core_verify_server_certificates(LinphoneCore *lc, bool_t yesno){
sal_verify_server_certificates(lc->sal,yesno);
......@@ -3718,6 +3747,7 @@ void linphone_core_verify_server_certificates(LinphoneCore *lc, bool_t yesno){
/**
* Specify whether the tls server certificate common name must be verified when connecting to a SIP/TLS server.
* @ingroup initializing
**/
void linphone_core_verify_server_cn(LinphoneCore *lc, bool_t yesno){
sal_verify_server_cn(lc->sal,yesno);
......@@ -4641,6 +4671,12 @@ void *linphone_core_get_user_data(LinphoneCore *lc){
return lc->data;
}
/**
* Associate a user pointer to the linphone core.
*
* @ingroup initializing
**/
void linphone_core_set_user_data(LinphoneCore *lc, void *userdata){
lc->data=userdata;
}
......@@ -4649,6 +4685,13 @@ int linphone_core_get_mtu(const LinphoneCore *lc){
return lc->net_conf.mtu;
}
/**
* Sets the maximum transmission unit size in bytes.
* This information is useful for sending RTP packets.
* Default value is 1500.
*
* @ingroup media_parameters
**/
void linphone_core_set_mtu(LinphoneCore *lc, int mtu){
lc->net_conf.mtu=mtu;
if (mtu>0){
......
......@@ -1006,16 +1006,8 @@ int linphone_core_get_upload_bandwidth(const LinphoneCore *lc);
void linphone_core_enable_adaptive_rate_control(LinphoneCore *lc, bool_t enabled);
bool_t linphone_core_adaptive_rate_control_enabled(const LinphoneCore *lc);
/**
* set audio packetization time linphone expect to receive from peer
* @ingroup media_parameters
*
*/
void linphone_core_set_download_ptime(LinphoneCore *lc, int ptime);
/**
* get audio packetization time linphone expect to receive from peer, 0 means unspecified
* @ingroup media_parameters
*/
int linphone_core_get_download_ptime(LinphoneCore *lc);
void linphone_core_set_upload_ptime(LinphoneCore *lc, int ptime);
......@@ -1046,7 +1038,7 @@ int linphone_core_enable_payload_type(LinphoneCore *lc, PayloadType *pt, bool_t
*/
#define LINPHONE_FIND_PAYLOAD_IGNORE_CHANNELS -1
/**
* Get payload type from mime type and clock rate
* Get payload type from mime type and clock rate
* @ingroup media_parameters
* This function searches in audio and video codecs for the given payload type name and clockrate.
* @param lc #LinphoneCore object
......
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