streamMedia_interface.h

    1 
    2 
    3 /*
    4  * ====================== WARNING ======================
    5  *
    6  * THE CONTENTS OF THIS FILE HAVE BEEN AUTO-GENERATED.
    7  * DO NOT MODIFY IN ANY WAY.
    8  *
    9  * ====================== WARNING ======================
   10  */
   11 
   12 /**
   13  * @page c_streamMedia Stream Media API
   14  *
   15  * @ref streamMedia_interface.h "API Reference" <br>
   16  * <HR>
   17  *
   18  * @warning This interface is experimental and unsupported.
   19  *
   20  * The Stream Media API handles media streaming interfaces with real-time protocols like RTP.
   21  *
   22  * The Real-time Transport Protocol is used to stream audio over IP/Ethernet. It is used to
   23  * transport any audio stream through an IP or Ethernet network.
   24  *
   25  * The RTP protocol is augmented by a control protocol called Real-Time Transport Control Protocol
   26  * (RTCP), which provides monitoring capabilities on the data delivery and minimum control
   27  * functionality.
   28  *
   29  * More information on the protocol can be found in
   30  * <a href="https://tools.ietf.org/pdf/rfc3550.pdf">RFC-3550</a>. A description of the RTP/AVP
   31  * profile, which describes generic audio and video conferences, can be found in
   32  * <a href="https://tools.ietf.org/pdf/rfc3551.pdf">RFC-3551</a>.
   33  *
   34  * @note The RTP interface does not provide the full Audio/Video Profile (AVP) profile : it can only
   35  * handle a singlecast audio session.
   36  *
   37  * Typically, the RTP interface can be used to redirect an audio call coming from an audio interface
   38  * through an IP network. For example, an IP audio server (Legato platform) would retrieve audio
   39  * from any audio interface (e.g. a Modem voice call), and connect it to the RTP interface that
   40  * sends the data through an IP network. The data would be received by an IP client (PC, smartphone,
   41  * Legato platform) that is able to decode the audio stream.
   42  *
   43  * @image html AudioOverIP.png
   44  *
   45  * In order to start a RTP session, information on the local and remote IP address are required to
   46  * set up UDP sockets. The local port corresponds to the UDP socket on which RTP packets will be
   47  * sent and received, and the remote address and port correspond to the IP address and port on which
   48  * RTP packets should be sent.
   49  *
   50  * @section streamMedia_binding IPC interfaces binding
   51  *
   52  * All the functions of this API are provided by the @b streamMedia sample service.
   53  *
   54  * Here's a code sample binding to streamMedia service:
   55  * @verbatim
   56    bindings:
   57    {
   58       clientExe.clientComponent.streamMedia -> streamMedia.streamMedia
   59    }
   60    @endverbatim
   61  *
   62  * @section streamMedia_interfaces Open/Close an Audio Interface
   63  *
   64  * The following functions let you select the desired interface attributes:
   65  * - streamMedia_OpenAudioRtpRx(): return an Audio Stream Reference of the digitized audio signal
   66  *                                 coming from a RTP socket.
   67  * - streamMedia_OpenAudioRtpTx(): return an Audio Stream Reference of the digitized audio signal
   68  *                                 routed to a RTP socket.
   69  *
   70  * @section streamMedia_rtp RTP
   71  *
   72  * The streamMedia_OpenAudioRtpRx() and streamMedia_OpenAudioRtpTx() functions allow the
   73  * application to set up a RTP session.
   74  *
   75  * After opening and connecting a RTP interface, it has to be started using streamMedia_Start().
   76  *
   77  * You can also register a handler function for RTCP packet reception management.
   78  *
   79  * streamMedia_AddRtcpHandler() function installs a handler for RTCP Packet reception.
   80  *
   81  * streamMedia_RemoveRtcpHandler() function removes the RTCP handler function.
   82  *
   83  * The streamMedia_SendRtcpSdes() and streamMedia_SendRtcpBye() functions allow the
   84  * application to send RTCP Session Description (SDES) or BYE packet.
   85  *
   86  * The RTCP SDES packet can be used to mark the beginning of the RTP session. It is composed of
   87  * several optional text fields that can provide information about the user's name, email address,
   88  * etc... See streamMedia_SendRtcpSdes() for more details.
   89  *
   90  * The RTCP BYE packet can be used to mark the end of the RTP session. It is possible to specify the
   91  * reason. See streamMedia_SendRtcpBye() for more details.
   92  *
   93  * More information on the RTCP SDES and BYE packets can be found in
   94  * <a href="https://tools.ietf.org/pdf/rfc3550.pdf">RFC-3550</a>.
   95  *
   96  * Before disconnecting and closing a RTP interface, it has to be stopped using
   97  * streamMedia_Stop(). Then, after disconnecting the interface from the connector,
   98  * streamMedia_Close() is called to close it.
   99  *
  100  * @note The RTP interface requires one more UDP socket in order to send and receive RTCP packets.
  101  * The RTCP UDP socket port number is automatically set to the local RTP port plus one.
  102  *
  103  * @section streamMedia_code Sample code
  104  *
  105  * The following samples illustrate the case described in the image above. It consists of two
  106  * different devices that communicate through RTP over an IP network. One device handles a modem
  107  * voice call, and the other one handles the microphone and speaker.
  108  *
  109  * @warning The following codes show a basic example of how to transmit audio through RTP between
  110  * two peers. Quality of service is not provided in those samples. They just show how the
  111  * streamMedia API should work and how it should be used.
  112  *
  113  * This first sample code is related to the RTP peer that handles the modem voice call.
  114  *
  115  * @code
  116  *
  117  * // Peer 1 : connection between a modem voice call and a RTP session.
  118  *
  119  * static void MyCallEventHandler
  120  * (
  121  *     le_mcc_CallRef_t   callRef,
  122  *     le_mcc_Event_t     callEvent,
  123  *     void*              contextPtr
  124  * )
  125  * {
  126  *     if (callEvent == LE_MCC_EVENT_TERMINATED)
  127  *     {
  128  *         le_mcc_TerminationReason_t term = le_mcc_GetTerminationReason(callRef);
  129  *         le_mcc_Delete(callRef);
  130  *         DisconnectAllAudio(); // Not showed here, would disconnect all audio.
  131  *     }
  132  *     else if (callEvent == LE_MCC_EVENT_INCOMING)
  133  *     {
  134  *         le_mcc_Answer(callRef);
  135  *     }
  136  *     else
  137  *     {
  138  *         LE_INFO("Other Call event.%d", callEvent);
  139  *     }
  140  * }
  141  *
  142  * static void MyRtcpEventHandler
  143  * (
  144  *     le_audio_StreamRef_t          streamRef,
  145  *     streamMedia_RtcpEvent_t    event,
  146  *     void*                         contextPtr
  147  * )
  148  * {
  149  *     switch (event)
  150  *     {
  151  *         case STREAMMEDIA_RTCP_BYE:
  152  *             DisconnectAllAudio(); // Not showed here, would disconnect all audio.
  153  *             break;
  154  *         default:
  155  *             LE_INFO("Other event");
  156  *             break;
  157  *     }
  158  * }
  159  *
  160  * void ConnectModemToRtp
  161  * (
  162  *     void
  163  * )
  164  * {
  165  *     static le_audio_ConnectorRef_t  audioInputConnectorRef = NULL;
  166  *     static le_audio_ConnectorRef_t  audioOutputConnectorRef = NULL;
  167  *     static le_audio_StreamRef_t     rtpInRef = NULL;
  168  *     static le_audio_StreamRef_t     rtpOutRef = NULL;
  169  *     static le_audio_StreamRef_t     mdmRxAudioRef = NULL;
  170  *     static le_audio_StreamRef_t     mdmTxAudioRef = NULL;
  171  *     static le_mcc_CallRef_t         callRef = NULL;
  172  *     const char*                     remoteAddress = "10.40.58.2";
  173  *     const char*                     destinationNumber = "0123456789";
  174  *     int                             localPort = 4000;
  175  *     int                             remotePort = 4000;
  176  *
  177  *     rtpInRef = streamMedia_OpenAudioRtpRx(localPort);
  178  *     rtpOutRef = streamMedia_OpenAudioRtpTx(localPort, remoteAddress, remotePort);
  179  *
  180  *     mdmRxAudioRef = le_audio_OpenModemVoiceRx();
  181  *     mdmTxAudioRef = le_audio_OpenModemVoiceTx();
  182  *
  183  *     audioInputConnectorRef = le_audio_CreateConnector();
  184  *     audioOutputConnectorRef = le_audio_CreateConnector();
  185  *
  186  *     le_audio_Connect(audioOutputConnectorRef, mdmTxAudioRef);
  187  *     le_audio_Connect(audioInputConnectorRef, mdmRxAudioRef);
  188  *
  189  *     le_audio_Connect(audioOutputConnectorRef, rtpInRef);
  190  *     streamMedia_Start(rtpInRef);
  191  *
  192  *     le_audio_Connect(audioInputConnectorRef, rtpOutRef);
  193  *     streamMedia_Start(rtpOutRef);
  194  *
  195  *     streamMedia_AddRtcpHandler(rtpInRef, MyRtcpEventHandler, NULL);
  196  *
  197  *     le_mcc_AddCallEventHandler(MyCallEventHandler, NULL);
  198  *     callRef=le_mcc_Create(destinationNumber);
  199  *     le_mcc_Start(callRef);
  200  * }
  201  *
  202  * @endcode
  203  *
  204  * This sample code shows how to connect RTP to microphone and speaker.
  205  *
  206  * @code
  207  *
  208  * // Peer 2 : connection between a RTP session and microphone and speaker.
  209  *
  210  * static void MyRtcpEventHandler
  211  * (
  212  *     le_audio_StreamRef_t          streamRef,
  213  *     streamMedia_RtcpEvent_t    event,
  214  *     void*                         contextPtr
  215  * )
  216  * {
  217  *     switch (event)
  218  *     {
  219  *         case STREAMMEDIA_RTCP_BYE:
  220  *             DisconnectAllAudio(); // Not showed here, would disconnect all audio.
  221  *             break;
  222  *         default:
  223  *             LE_INFO("Other event");
  224  *             break;
  225  *     }
  226  * }
  227  *
  228  * void RecordFileFromRtp
  229  * (
  230  *     void
  231  * )
  232  * {
  233  *     static le_audio_ConnectorRef_t  audioInputConnectorRef = NULL;
  234  *     static le_audio_ConnectorRef_t  audioOutputConnectorRef = NULL;
  235  *     static le_audio_StreamRef_t     rtpInRef = NULL;
  236  *     static le_audio_StreamRef_t     rtpOutRef = NULL;
  237  *     static le_audio_StreamRef_t     micAudioRef = NULL;
  238  *     static le_audio_StreamRef_t     speakerAudioRef = NULL;
  239  *     const char*                     remoteAddress = "10.40.58.2";
  240  *     int                             localPort = 4000;
  241  *     int                             remotePort = 4000;
  242  *
  243  *     rtpInRef = streamMedia_OpenAudioRtpRx(localPort);
  244  *     rtpOutRef = streamMedia_OpenAudioRtpTx(localPort, remoteAddress, remotePort);
  245  *
  246  *     micAudioRef = le_audio_OpenMic();
  247  *     speakerAudioRef = le_audio_OpenSpeaker();
  248  *
  249  *     audioInputConnectorRef = le_audio_CreateConnector();
  250  *     audioOutputConnectorRef = le_audio_CreateConnector();
  251  *
  252  *     le_audio_Connect(audioOutputConnectorRef, speakerAudioRef);
  253  *     le_audio_Connect(audioInputConnectorRef, micAudioRef);
  254  *
  255  *     le_audio_Connect(audioOutputConnectorRef, rtpInRef);
  256  *     streamMedia_Start(rtpInRef);
  257  *
  258  *     le_audio_Connect(audioInputConnectorRef, rtpOutRef);
  259  *     streamMedia_Start(rtpOutRef);
  260  *
  261  *     streamMedia_AddRtcpHandler(rtpInRef, MyRtcpEventHandler, NULL);
  262  * }
  263  *
  264  * @endcode
  265  *
  266  * @section streamMedia_warning Known limitations
  267  *
  268  * The RTP interface must be connected and disconnected in this specific order :
  269  * - First connect the RTP reception stream, and then the transmission one.
  270  * - When disconnecting it, start by the transmission stream, and then disconnect the reception one.
  271  *
  272  * The streamMedia API provided by the streamMedia sample application is not suitable for
  273  * real-time communication applications because of the too high latency.
  274  * A delay of 500 ms has been measured on a one-sided RTP session (RTP transmission only) on a WP85
  275  * module. Thus, the "Audio over IP" use case described above shows a latency of more than one
  276  * second due to the two-sided RTP session (transmission on one module and reception on the other).
  277  * However, the streamMedia API can be used for any streaming purposes where a delay of one second
  278  * is acceptable.
  279  *
  280  * The streamMedia application has been tested on WP85 and AR8652.
  281  *
  282  * <HR>
  283  *
  284  * Copyright (C) Sierra Wireless Inc.
  285  */
  286 /**
  287  * @file streamMedia_interface.h
  288  *
  289  * Legato @ref sampleApps_streamMedia include file.
  290  *
  291  * Copyright (C) Sierra Wireless Inc.
  292  */
  293 
  294 #ifndef STREAMMEDIA_INTERFACE_H_INCLUDE_GUARD
  295 #define STREAMMEDIA_INTERFACE_H_INCLUDE_GUARD
  296 
  297 
  298 #include "legato.h"
  299 
  300 // Interface specific includes
  301 #include "le_audio_interface.h"
  302 
  303 // Internal includes for this interface
  304 #include "streamMedia_common.h"
  305 //--------------------------------------------------------------------------------------------------
  306 /**
  307  * Type for handler called when a server disconnects.
  308  */
  309 //--------------------------------------------------------------------------------------------------
  310 typedef void (*streamMedia_DisconnectHandler_t)(void *);
  311 
  312 //--------------------------------------------------------------------------------------------------
  313 /**
  314  *
  315  * Connect the current client thread to the service providing this API. Block until the service is
  316  * available.
  317  *
  318  * For each thread that wants to use this API, either ConnectService or TryConnectService must be
  319  * called before any other functions in this API.  Normally, ConnectService is automatically called
  320  * for the main thread, but not for any other thread. For details, see @ref apiFilesC_client.
  321  *
  322  * This function is created automatically.
  323  */
  324 //--------------------------------------------------------------------------------------------------
  325 void streamMedia_ConnectService
  326 (
  327     void
  328 );
  329 
  330 //--------------------------------------------------------------------------------------------------
  331 /**
  332  *
  333  * Try to connect the current client thread to the service providing this API. Return with an error
  334  * if the service is not available.
  335  *
  336  * For each thread that wants to use this API, either ConnectService or TryConnectService must be
  337  * called before any other functions in this API.  Normally, ConnectService is automatically called
  338  * for the main thread, but not for any other thread. For details, see @ref apiFilesC_client.
  339  *
  340  * This function is created automatically.
  341  *
  342  * @return
  343  *  - LE_OK if the client connected successfully to the service.
  344  *  - LE_UNAVAILABLE if the server is not currently offering the service to which the client is
  345  *    bound.
  346  *  - LE_NOT_PERMITTED if the client interface is not bound to any service (doesn't have a binding).
  347  *  - LE_COMM_ERROR if the Service Directory cannot be reached.
  348  */
  349 //--------------------------------------------------------------------------------------------------
  350 le_result_t streamMedia_TryConnectService
  351 (
  352     void
  353 );
  354 
  355 //--------------------------------------------------------------------------------------------------
  356 /**
  357  * Set handler called when server disconnection is detected.
  358  *
  359  * When a server connection is lost, call this handler then exit with LE_FATAL.  If a program wants
  360  * to continue without exiting, it should call longjmp() from inside the handler.
  361  */
  362 //--------------------------------------------------------------------------------------------------
  363 LE_FULL_API void streamMedia_SetServerDisconnectHandler
  364 (
  365     streamMedia_DisconnectHandler_t disconnectHandler,
  366     void *contextPtr
  367 );
  368 
  369 //--------------------------------------------------------------------------------------------------
  370 /**
  371  *
  372  * Disconnect the current client thread from the service providing this API.
  373  *
  374  * Normally, this function doesn't need to be called. After this function is called, there's no
  375  * longer a connection to the service, and the functions in this API can't be used. For details, see
  376  * @ref apiFilesC_client.
  377  *
  378  * This function is created automatically.
  379  */
  380 //--------------------------------------------------------------------------------------------------
  381 void streamMedia_DisconnectService
  382 (
  383     void
  384 );
  385 
  386 
  387 //--------------------------------------------------------------------------------------------------
  388 /**
  389  * RTCP packet reception events, related to the RTCP packet type. See
  390  * <a href="https://tools.ietf.org/pdf/rfc3550.pdf">RFC-3550</a> for more details.
  391  */
  392 //--------------------------------------------------------------------------------------------------
  393 
  394 
  395 //--------------------------------------------------------------------------------------------------
  396 /**
  397  * Handler for RTCP packet reception.
  398  *
  399  */
  400 //--------------------------------------------------------------------------------------------------
  401 
  402 
  403 //--------------------------------------------------------------------------------------------------
  404 /**
  405  * Reference type used by Add/Remove functions for EVENT 'streamMedia_Rtcp'
  406  */
  407 //--------------------------------------------------------------------------------------------------
  408 
  409 
  410 //--------------------------------------------------------------------------------------------------
  411 /**
  412  * Open the received audio stream of the RTP interface.
  413  *
  414  * @return Reference to the input audio stream, NULL if the function fails.
  415  */
  416 //--------------------------------------------------------------------------------------------------
  417 le_audio_StreamRef_t streamMedia_OpenAudioRtpRx
  418 (
  419     int32_t localPort
  420         ///< [IN] The local UDP socket port for the RTP session.
  421 );
  422 
  423 //--------------------------------------------------------------------------------------------------
  424 /**
  425  * Open the transmitted audio stream of the RTP interface.
  426  *
  427  * @return Reference to the output audio stream, NULL if the function fails.
  428  */
  429 //--------------------------------------------------------------------------------------------------
  430 le_audio_StreamRef_t streamMedia_OpenAudioRtpTx
  431 (
  432     int32_t localPort,
  433         ///< [IN] The local UDP socket port for the RTP
  434         ///< session.
  435     const char* LE_NONNULL remoteIpAddress,
  436         ///< [IN] The IP address of the remote peer.
  437     int32_t remotePort
  438         ///< [IN] The port of the remote peer.
  439 );
  440 
  441 //--------------------------------------------------------------------------------------------------
  442 /**
  443  * Start the RTP stream. This must be called after connecting the RTP interface to start
  444  * RTP reception or transmission.
  445  *
  446  * @return LE_FAULT         Function failed.
  447  * @return LE_OK            Function succeeded.
  448  */
  449 //--------------------------------------------------------------------------------------------------
  450 le_result_t streamMedia_Start
  451 (
  452     le_audio_StreamRef_t streamRef
  453         ///< [IN] Audio stream reference.
  454 );
  455 
  456 //--------------------------------------------------------------------------------------------------
  457 /**
  458  * Stop the RTP stream.
  459  *
  460  * @return LE_FAULT         Function failed.
  461  * @return LE_OK            Function succeeded.
  462  */
  463 //--------------------------------------------------------------------------------------------------
  464 le_result_t streamMedia_Stop
  465 (
  466     le_audio_StreamRef_t streamRef
  467         ///< [IN] Audio stream reference.
  468 );
  469 
  470 //--------------------------------------------------------------------------------------------------
  471 /**
  472  * Send a RTCP Session Description packet (SDES) using
  473  * <a href="http://www.pjsip.org/pjmedia/docs/html/group__PJMED__RTCP.htm#
  474  * ga8ddca87e2b3ab0b02635edd66d7de748">pjmedia_rtcp_build_rtcp_sdes()</a>.
  475  *
  476  * @return LE_FAULT         Function failed.
  477  * @return LE_OK            Function succeeded.
  478  */
  479 //--------------------------------------------------------------------------------------------------
  480 le_result_t streamMedia_SendRtcpSdes
  481 (
  482     le_audio_StreamRef_t streamRef,
  483         ///< [IN] Audio stream reference.
  484     const char* LE_NONNULL cname,
  485         ///< [IN] The optional source canonical name.
  486     const char* LE_NONNULL name,
  487         ///< [IN] The optional source name.
  488     const char* LE_NONNULL email,
  489         ///< [IN] The optional source email.
  490     const char* LE_NONNULL phone,
  491         ///< [IN] The optional source phone.
  492     const char* LE_NONNULL loc,
  493         ///< [IN] The optional source location.
  494     const char* LE_NONNULL tool,
  495         ///< [IN] The optional source tool.
  496     const char* LE_NONNULL note
  497         ///< [IN] The optional source note.
  498 );
  499 
  500 //--------------------------------------------------------------------------------------------------
  501 /**
  502  * Send a RTCP BYE packet using
  503  * <a href="http://www.pjsip.org/pjmedia/docs/html/group__PJMED__RTCP.htm#
  504  * ga9eb25597d5815cee68fe7fdc3b4cd9e9">pjmedia_rtcp_build_rtcp_bye()</a>.
  505  *
  506  * @return LE_FAULT         Function failed.
  507  * @return LE_OK            Function succeeded.
  508  */
  509 //--------------------------------------------------------------------------------------------------
  510 le_result_t streamMedia_SendRtcpBye
  511 (
  512     le_audio_StreamRef_t streamRef,
  513         ///< [IN] Audio stream reference.
  514     const char* LE_NONNULL reason
  515         ///< [IN] The optional BYE reason.
  516 );
  517 
  518 //--------------------------------------------------------------------------------------------------
  519 /**
  520  * Close an RTP stream.
  521  *
  522  */
  523 //--------------------------------------------------------------------------------------------------
  524 void streamMedia_Close
  525 (
  526     le_audio_StreamRef_t streamRef
  527         ///< [IN] Audio stream reference.
  528 );
  529 
  530 //--------------------------------------------------------------------------------------------------
  531 /**
  532  * Add handler function for EVENT 'streamMedia_Rtcp'
  533  *
  534  * This event provides information on RTCP packet type.
  535  *
  536  */
  537 //--------------------------------------------------------------------------------------------------
  538 streamMedia_RtcpHandlerRef_t streamMedia_AddRtcpHandler
  539 (
  540     le_audio_StreamRef_t streamRef,
  541         ///< [IN] Audio stream reference.
  542     streamMedia_RtcpHandlerFunc_t handlerPtr,
  543         ///< [IN]
  544     void* contextPtr
  545         ///< [IN]
  546 );
  547 
  548 //--------------------------------------------------------------------------------------------------
  549 /**
  550  * Remove handler function for EVENT 'streamMedia_Rtcp'
  551  */
  552 //--------------------------------------------------------------------------------------------------
  553 void streamMedia_RemoveRtcpHandler
  554 (
  555     streamMedia_RtcpHandlerRef_t handlerRef
  556         ///< [IN]
  557 );
  558 
  559 #endif // STREAMMEDIA_INTERFACE_H_INCLUDE_GUARD
streamMedia_SendRtcpBye
le_result_t streamMedia_SendRtcpBye(le_audio_StreamRef_t streamRef, const char *LE_NONNULL reason)
le_result_t
le_result_t
Definition: le_basics.h:45
streamMedia_DisconnectHandler_t
void(* streamMedia_DisconnectHandler_t)(void *)
Definition: streamMedia_interface.h:310
streamMedia_OpenAudioRtpRx
le_audio_StreamRef_t streamMedia_OpenAudioRtpRx(int32_t localPort)
streamMedia_AddRtcpHandler
streamMedia_RtcpHandlerRef_t streamMedia_AddRtcpHandler(le_audio_StreamRef_t streamRef, streamMedia_RtcpHandlerFunc_t handlerPtr, void *contextPtr)
streamMedia_Stop
le_result_t streamMedia_Stop(le_audio_StreamRef_t streamRef)
streamMedia_DisconnectService
void streamMedia_DisconnectService(void)
le_audio_interface.h
LE_FULL_API
#define LE_FULL_API
Definition: le_apiFeatures.h:40
streamMedia_Start
le_result_t streamMedia_Start(le_audio_StreamRef_t streamRef)
streamMedia_SetServerDisconnectHandler
LE_FULL_API void streamMedia_SetServerDisconnectHandler(streamMedia_DisconnectHandler_t disconnectHandler, void *contextPtr)
streamMedia_RemoveRtcpHandler
void streamMedia_RemoveRtcpHandler(streamMedia_RtcpHandlerRef_t handlerRef)
streamMedia_OpenAudioRtpTx
le_audio_StreamRef_t streamMedia_OpenAudioRtpTx(int32_t localPort, const char *LE_NONNULL remoteIpAddress, int32_t remotePort)
streamMedia_TryConnectService
le_result_t streamMedia_TryConnectService(void)
streamMedia_ConnectService
void streamMedia_ConnectService(void)
streamMedia_Close
void streamMedia_Close(le_audio_StreamRef_t streamRef)
streamMedia_SendRtcpSdes
le_result_t streamMedia_SendRtcpSdes(le_audio_StreamRef_t streamRef, const char *LE_NONNULL cname, const char *LE_NONNULL name, const char *LE_NONNULL email, const char *LE_NONNULL phone, const char *LE_NONNULL loc, const char *LE_NONNULL tool, const char *LE_NONNULL note)