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
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
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"
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);
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.