commit eb7def234e2fc6fd16cc627eaef813d2316c6ed6
author fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Dec 09 21:34:30 2013
committer fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Dec 09 21:34:30 2013
tree 1c04dc26a7b458eff35ec0e04e28eaa4c563c207
parent c329529047298128b1c6f472ee49b0b134b10f13

Fix compilation errors on Fedora 20.

peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.

BUG=2700
R=andrew@webrtc.org, fischman@webrtc.org, henrik.lundin@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5019004

Patch from Victor Costan <costan@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d

AUTHORS

diff --git a/AUTHORS b/AUTHORS
index 5fe488f..e5f2839 100644
--- a/AUTHORS
+++ b/AUTHORS
@@ -10,6 +10,7 @@
 Pali Rohar
 Robert Nagy
 Silviu Caragea <silviu.cpp@gmail.com>
+Victor Costan <costan@gmail.com>
 
 Google Inc.
 Intel Corporation

talk/app/webrtc/java/jni/peerconnection_jni.cc

diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index 0b7c78b..75f230f 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -61,6 +61,7 @@
 #include <map>
 #include <sys/prctl.h>
 #include <sys/syscall.h>
+#include <unistd.h>
 
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"

webrtc/modules/audio_coding/neteq4/test/RTPtimeshift.cc

diff --git a/webrtc/modules/audio_coding/neteq4/test/RTPtimeshift.cc b/webrtc/modules/audio_coding/neteq4/test/RTPtimeshift.cc
index ba3a08e..15ffdf6 100644
--- a/webrtc/modules/audio_coding/neteq4/test/RTPtimeshift.cc
+++ b/webrtc/modules/audio_coding/neteq4/test/RTPtimeshift.cc
@@ -68,18 +68,20 @@
 	uint32_t ATdiff = 0;
     if (argc > 4)
     {
-        if (argv[4] >= 0)
-            SNdiff = atoi(argv[4]) - packet.sequenceNumber();
+        int startSN = atoi(argv[4]);
+        if (startSN >= 0)
+            SNdiff = startSN - packet.sequenceNumber();
         if (argc > 5)
         {
-            if (argv[5] >= 0)
-                ATdiff = atoi(argv[5]) - packet.time();
+            int startTS = atoi(argv[5]);
+            if (startTS >= 0)
+                ATdiff = startTS - packet.time();
         }
     }
 
     while (packLen >= 0)
     {
-        
+
         packet.setTimeStamp(packet.timeStamp() + TSdiff);
         packet.setSequenceNumber(packet.sequenceNumber() + SNdiff);
         packet.setTime(packet.time() + ATdiff);

webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index 8ef1074..96fa80a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -38,7 +38,7 @@
 
  protected:
   ModuleRTPUtility::Payload* ExpectReturnOfTypicalAudioPayload(
-      uint8_t payload_type, int rate) {
+      uint8_t payload_type, uint32_t rate) {
     bool audio = true;
     ModuleRTPUtility::Payload returned_payload = { "name", audio, {
         // Initialize the audio struct in this case.