From eb7def234e2fc6fd16cc627eaef813d2316c6ed6 Mon Sep 17 00:00:00 2001 From: "fischman@webrtc.org" Date: Mon, 9 Dec 2013 21:34:30 +0000 Subject: [PATCH] Fix compilation errors on Fedora 20. peerconnection_jni.cc: syscall() comes from 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 . git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d --- AUTHORS | 1 + talk/app/webrtc/java/jni/peerconnection_jni.cc | 1 + .../modules/audio_coding/neteq4/test/RTPtimeshift.cc | 12 +++++++----- .../rtp_rtcp/source/rtp_payload_registry_unittest.cc | 2 +- 4 files changed, 10 insertions(+), 6 deletions(-) diff --git a/AUTHORS b/AUTHORS index 5fe488f26..e5f2839ba 100644 --- a/AUTHORS +++ b/AUTHORS @@ -10,6 +10,7 @@ Martin Storsjo Pali Rohar Robert Nagy Silviu Caragea +Victor Costan 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 0b7c78bc1..75f230f92 100644 --- a/talk/app/webrtc/java/jni/peerconnection_jni.cc +++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc @@ -61,6 +61,7 @@ #include #include #include +#include #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 ba3a08ee0..15ffdf6a5 100644 --- a/webrtc/modules/audio_coding/neteq4/test/RTPtimeshift.cc +++ b/webrtc/modules/audio_coding/neteq4/test/RTPtimeshift.cc @@ -68,18 +68,20 @@ int main(int argc, char* argv[]) 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 8ef10741f..96fa80ad8 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 @@ class RtpPayloadRegistryTest : public ::testing::Test { 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.