[asterisk-scf-commits] asterisk-scf/integration/slice.git branch "rtcp" updated.

[Commits to the Asterisk SCF project code repositories]

branch "rtcp" has been updated
       via  343c362d36600da519491e244e296d53f5480096 (commit)
      from  28ee15919bebb57c53a7cc622c636c8822fba277 (commit)

Summary of changes:
 slice/AsteriskSCF/Media/RTP/MediaRTCPIf.ice |   58 +++++++++++++++++---------
 1 files changed, 38 insertions(+), 20 deletions(-)


- Log -----------------------------------------------------------------
commit 343c362d36600da519491e244e296d53f5480096
Author: Joshua Colp <jcolp at digium.com>
Date:   Mon Jul 11 13:51:13 2011 -0300

    Incorporate review feedback and make some additional tweaks.

diff --git a/slice/AsteriskSCF/Media/RTP/MediaRTCPIf.ice b/slice/AsteriskSCF/Media/RTP/MediaRTCPIf.ice
index ce06f03..eb591cc 100644
--- a/slice/AsteriskSCF/Media/RTP/MediaRTCPIf.ice
+++ b/slice/AsteriskSCF/Media/RTP/MediaRTCPIf.ice
@@ -14,6 +14,8 @@
  * at the top of the source tree.
  */
 
+#include <AsteriskSCF/Media/RTP/MediaRTPIf.ice>
+
 module AsteriskSCF
 {
 
@@ -34,7 +36,7 @@ module V1
    /**
     * Fine grained RTCP statistics information for certain aspects
     */
-   class RTCPExtendedDetails
+   class ExtendedDetails
    {
        /**
         * Maximum value
@@ -58,16 +60,14 @@ module V1
    };
 
    /**
-    * Concrete class containing RTCP statistics information. If retrieved from a proxy
-    * to a sink the statistics will be for the sending side. If retrieved from a proxy
-    * to a source the statistics will be for the receiving side.
+    * Concrete class containing RTCP statistics information.
     */
-   class RTCPStatistics
+   class Statistics
    {
        /**
         * Round trip delay information (values are in usec)
         */
-       RTCPExtendedDetails roundTripDelay;
+       ExtendedDetails roundTripDelay;
 
        /**
         * Number of packets in total
@@ -97,18 +97,18 @@ module V1
        /**
         * Packet loss information (values are in usec)
         */
-       RTCPExtendedDetails loss;
+       ExtendedDetails loss;
 
        /**
         * Jitter information (values are in usec)
         */
-       RTCPExtendedDetails jitter;
+       ExtendedDetails jitter;
    };
 
    /**
     * Facet name for the RTCP session interface
     */
-   const string RTCPSessionFacet = "rtcpsession";
+   const string SessionFacet = "rtcpsession";
 
    /**
     * Interface to an RTCP session
@@ -120,7 +120,7 @@ module V1
         *
         * @return int The local port for our RTCP session
         */
-       int getLocalPort();
+       idempotent int getLocalPort();
 
        /**
         * Method used to set the port that RTCP should be sent to
@@ -135,54 +135,72 @@ module V1
    /**
     * An RTCP information listener interface
     */
-   interface RTCPInformationListener
+   interface InformationListener
    {
        /**
         * Method called when statistics information is updated for a source
         *
-        * @param statistics Updated statistics information
+        * @param source The source the statistics were updated on
+        *
+        * @param newStatistics Updated statistics information
         */
-       void sourceStatisticsUpdated(RTCPStatistics statistics);
+       void sourceStatisticsUpdated(AsteriskSCF::Media::RTP::V1::StreamSourceRTP* source, Statistics newStatistics);
 
        /**
         * Method called when statistics information is updated for a sink
         *
-        * @param statistics Updated statistics information
+        *
+        * @param sink The sink the statistics were updated on
+        *
+        * @param newStatistics Updated statistics information
+        */
+       void sinkStatisticsUpdated(AsteriskSCF::Media::RTP::V1::StreamSinkRTP* sink, Statistics newStatistics);
+
+       /**
+        * Method called when the SSRC changes for a source
+        *
+        * @param source The source that the SSRC has changed on
+        *
+        * @param ssrc The new SSRC value
         */
-       void sinkStatisticsUpdated(RTCPStatistics statistics);
+       void sourceSsrcChanged(AsteriskSCF::Media::RTP::V1::StreamSourceRTP* source, int ssrc);
    };
 
    /**
     * Facet name for the RTCP interface
     */
-   const string RTCPFacet = "rtcp";
+   const string Facet = "rtcp";
 
    /**
     * An RTCP statistics information interface. This is made available on media sources and sinks originating from
     * RTP sessions as a facet with the above name.
+    *
+    * If retrieved from a source the statistics returned will be for media origiating from the remote party.
+    *
+    * If retrieved from a sink the statistics returned will be for media destined for the remote party.
     */
-   interface RTCPInformation
+   interface Information
    {
        /**
         * Method used to retrieve statistics information
         *
         * @return RTCPStatistics A populated RTPStatistics concrete class
         */
-       idempotent RTCPStatistics getStatistics();
+       idempotent Statistics getStatistics();
 
        /**
         * Method used to add a listener which receives statistics information at the RTCP interval
         *
         * @param listener A proxy to the listener to add
         */
-       void addListener(RTCPInformationListener *listener);
+       void addListener(InformationListener *listener);
 
        /**
         * Method used to remove a listener
         *
         * @param listener A proxy to the listener to move
         */
-       void removeListener(RTCPInformationListener *listener);
+       void removeListener(InformationListener *listener);
    };
 
 }; /*  end module V1 */

-----------------------------------------------------------------------


-- 
asterisk-scf/integration/slice.git