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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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25
webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
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@ -14,31 +14,6 @@
namespace webrtc {
class FakeRtpRtcpClock : public Clock {
public:
FakeRtpRtcpClock() {
time_in_ms_ = 123456;
}
// Return a timestamp in milliseconds relative to some arbitrary
// source; the source is fixed for this clock.
virtual WebRtc_Word64 TimeInMilliseconds() {
return time_in_ms_;
}
virtual int64_t TimeInMicroseconds() {
return time_in_ms_ * 1000;
}
// Retrieve an NTP absolute timestamp.
virtual void CurrentNtp(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
secs = time_in_ms_ / 1000;
frac = (time_in_ms_ % 1000) * 4294967;
}
void IncrementTime(WebRtc_UWord32 time_increment_ms) {
time_in_ms_ += time_increment_ms;
}
private:
WebRtc_Word64 time_in_ms_;
};
// This class sends all its packet straight to the provided RtpRtcp module.
// with optional packet loss.
class LoopBackTransport : public webrtc::Transport {