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Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.

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TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1044004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-01-21 07:42:11 +00:00
parent a3c82bf667
commit a678a3baee
67 changed files with 367 additions and 544 deletions
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6
webrtc/modules/video_coding/main/test/test_callbacks.cc
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@ -13,9 +13,9 @@
#include <cmath>
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "modules/video_coding/main/source/tick_time_base.h"
#include "rtp_dump.h"
#include "test_macros.h"
#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
@ -204,7 +204,7 @@ VCMDecodeCompleteCallback::DecodedBytes()
return _decodedBytes;
}
RTPSendCompleteCallback::RTPSendCompleteCallback(TickTimeBase* clock,
RTPSendCompleteCallback::RTPSendCompleteCallback(Clock* clock,
const char* filename):
_clock(clock),
_sendCount(0),
@ -258,7 +258,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
bool transmitPacket = true;
transmitPacket = PacketLoss();
WebRtc_UWord64 now = _clock->MillisecondTimestamp();
int64_t now = _clock->TimeInMilliseconds();
// Insert outgoing packet into list
if (transmitPacket)
{