[asterisk-scf-commits] asterisk-scf/release/media_rtp_pjmedia.git branch "master" updated.

[Commits to the Asterisk SCF project code repositories]

branch "master" has been updated
       via  34d0513bcbe9db15aed100eb38635f7d7de7eb20 (commit)
      from  458c78b6a0b5ca95265fb2ff8ad1ebbf0c2228ac (commit)

Summary of changes:
 src/RTPSink.cpp                 |   15 +++++++++++----
 src/RTPSource.cpp               |   18 +++++++++++++-----
 src/RTPTelephonyEventSink.cpp   |    3 ++-
 src/RTPTelephonyEventSink.h     |    8 ++++++--
 src/RTPTelephonyEventSource.cpp |    2 +-
 src/RTPTelephonyEventSource.h   |   11 +++++++----
 6 files changed, 40 insertions(+), 17 deletions(-)


- Log -----------------------------------------------------------------
commit 34d0513bcbe9db15aed100eb38635f7d7de7eb20
Author: Brent Eagles <beagles at digium.com>
Date:   Fri Sep 16 14:25:22 2011 -0230

    Fix Winsock/Winsock2 build errors on Windows.

diff --git a/src/RTPSink.cpp b/src/RTPSink.cpp
index da5d45a..4f4eec6 100644
--- a/src/RTPSink.cpp
+++ b/src/RTPSink.cpp
@@ -14,18 +14,25 @@
  * at the top of the source tree.
  */
 
+//
+// It is annoying that #include <windows.h> pulls in the old version of winsock. The fix is to include
+// winsock2 first.
+//
+#ifdef _WIN32
+#include <WinSock2.h>
+#endif
+
 #include "RTPSink.h"
 #include "RtpStateReplicationIf.h"
 #include "RTPTelephonyEventSink.h"
 
-#include <pjlib.h>
-#include <pjmedia.h>
-
 #include <Ice/Ice.h>
 #include <IceUtil/UUID.h>
-
 #include <boost/asio/detail/socket_ops.hpp>
 
+#include <pjlib.h>
+#include <pjmedia.h>
+
 #include <AsteriskSCF/Media/MediaIf.h>
 #include <AsteriskSCF/Media/RTP/MediaRTPIf.h>
 #include <AsteriskSCF/System/Component/ReplicaIf.h>
diff --git a/src/RTPSource.cpp b/src/RTPSource.cpp
index b9bf4f1..ea81708 100644
--- a/src/RTPSource.cpp
+++ b/src/RTPSource.cpp
@@ -14,19 +14,27 @@
  * at the top of the source tree.
  */
 
+//
+// It is annoying that #include <windows.h> pulls in the old version of winsock. The fix is to include
+// winsock2 first.
+//
+#ifdef _WIN32
+#include <WinSock2.h>
+#endif
+
+#include "RTPTelephonyEventSource.h"
 #include "RTPSource.h"
 #include "RtpStateReplicationIf.h"
-#include "RTPTelephonyEventSource.h"
-
-#include <pjlib.h>
-#include <pjmedia.h>
 
 #include <Ice/Ice.h>
 #include <IceUtil/UUID.h>
 #include <IceUtil/Timer.h>
 
-#include <boost/thread.hpp>
 #include <boost/asio/detail/socket_ops.hpp>
+#include <boost/thread.hpp>
+
+#include <pjlib.h>
+#include <pjmedia.h>
 
 #include <AsteriskSCF/Media/MediaIf.h>
 #include <AsteriskSCF/Media/RTP/MediaRTPIf.h>
diff --git a/src/RTPTelephonyEventSink.cpp b/src/RTPTelephonyEventSink.cpp
index 82cf3a7..0e62dd2 100644
--- a/src/RTPTelephonyEventSink.cpp
+++ b/src/RTPTelephonyEventSink.cpp
@@ -18,6 +18,7 @@
 
 #include <AsteriskSCF/Media/Formats/OtherFormats.h>
 #include <IceUtil/UUID.h>
+#include <pjmedia.h>
 
 namespace
 {
@@ -184,7 +185,7 @@ void RTPTelephonyEventSink::getSource_async(
     cb->ice_response(mStateItem->source);
 }
 
-pj_uint8_t RTPTelephonyEventSink::translateDTMF(Ice::Byte signal)
+unsigned char RTPTelephonyEventSink::translateDTMF(Ice::Byte signal)
 {
     if (signal >= '0' && signal <= '9')
     {
diff --git a/src/RTPTelephonyEventSink.h b/src/RTPTelephonyEventSink.h
index 407c1b2..fe458a1 100644
--- a/src/RTPTelephonyEventSink.h
+++ b/src/RTPTelephonyEventSink.h
@@ -16,13 +16,17 @@
 
 #pragma once
 
-#include <pjmedia.h>
 #include <AsteriskSCF/Replication/MediaRTPPJMedia/RtpStateReplicationIf.h>
 #include <AsteriskSCF/SessionCommunications/TelephonyEventsIf.h>
 
 #include "PJMediaTransport.h"
 #include "SessionAdapter.h"
 
+//
+// Forward declarations for pjmedia.
+//
+struct pjmedia_rtp_session;
+
 class RTPTelephonyEventSink : public AsteriskSCF::SessionCommunications::V1::TelephonyEventSink
 {
 public:
@@ -64,7 +68,7 @@ private:
     /**
      * Translate DTMF from ASCII into its RFC 4733-designated payload value
      */
-    pj_uint8_t translateDTMF(Ice::Byte signal);
+    unsigned char translateDTMF(Ice::Byte signal);
 
     /**
      * Replicate state
diff --git a/src/RTPTelephonyEventSource.cpp b/src/RTPTelephonyEventSource.cpp
index 3a6631a..5024232 100644
--- a/src/RTPTelephonyEventSource.cpp
+++ b/src/RTPTelephonyEventSource.cpp
@@ -205,7 +205,7 @@ void RTPTelephonyEventSource::distributeToSinks(const TelephonyEventPtr& event)
 
 // This function does no bounds checking and assumes that whoever
 // calls it will not call into it with invalid input (i.e. event > 15)
-Ice::Byte RTPTelephonyEventSource::translateDTMF(pj_uint8_t event)
+Ice::Byte RTPTelephonyEventSource::translateDTMF(unsigned char event)
 {
     if (event < 10)
     {
diff --git a/src/RTPTelephonyEventSource.h b/src/RTPTelephonyEventSource.h
index cd7eb13..cb315a6 100644
--- a/src/RTPTelephonyEventSource.h
+++ b/src/RTPTelephonyEventSource.h
@@ -20,11 +20,14 @@
 
 #include <AsteriskSCF/SessionCommunications/TelephonyEventsIf.h>
 #include <AsteriskSCF/Replication/MediaRTPPJMedia/RtpStateReplicationIf.h>
-
-#include <pjmedia.h>
-
 #include "SessionAdapter.h"
 
+//
+// Forward declarations for pjmedia.
+//
+struct pjmedia_rtp_hdr;
+struct pjmedia_rtp_dtmf_event;
+
 class RTPTelephonyEventSource : public AsteriskSCF::SessionCommunications::V1::TelephonyEventSource
 {
 public:
@@ -93,7 +96,7 @@ private:
     /**
      * Translate DTMF from its RFC 4733 event payload to ASCII representation
      */
-    Ice::Byte translateDTMF(pj_uint8_t event);
+    Ice::Byte translateDTMF(unsigned char event);
 
     /**
      * Replicate state to replicas

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


-- 
asterisk-scf/release/media_rtp_pjmedia.git