Split WebRTC browser tests in getusermedia and peerconnection tests.
The content browsertests file for WebRTC was growing a bit large. I extracted the getUserMedia-focused tests into a test of its own and left the peerconnection tests. Also renamed all WebRTC content browsertests to use the WebRtc naming convention, which seems to be the most common in the production code. Note: this patch will require some filter renames on the WebRTC bots. R=tommi@chromium.org Review URL: https://codereview.chromium.org/131203005 git-svn-id: http://src.chromium.org/svn/trunk/src/build@245524 4ff67af0-8c30-449e-8e8b-ad334ec8d88c
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@ -67,16 +67,16 @@ DownloadContentTest.ResumeInterruptedDownloadBadPrecondition
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DownloadContentTest.ResumeWithDeletedFile
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# http://crbug.com/181069
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WebrtcBrowserTest.CanSetupLegacyCall
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WebrtcBrowserTest.CallWithDataOnly
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WebrtcBrowserTest.CallWithDataAndMedia
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WebrtcBrowserTest.CanMakeEmptyCallThenAddStreamsAndRenegotiate
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WebRtcBrowserTest.CanSetupLegacyCall
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WebRtcBrowserTest.CallWithDataOnly
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WebRtcBrowserTest.CallWithDataAndMedia
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WebRtcBrowserTest.CanMakeEmptyCallThenAddStreamsAndRenegotiate
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WebRTCInternalsBrowserTest.WithRealPeerConnectionCall
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WebRTCInternalsBrowserTest.NegotiateUnsupportedVideoCodec
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WebRTCInternalsBrowserTest.NegotiateNonCryptoCall
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# http://crbug.com/175683
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WebrtcBrowserTest.CallWithDataAndLaterAddMedia
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WebRtcBrowserTest.CallWithDataAndLaterAddMedia
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# http://crbug.com/224134
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RenderWidgetHostBrowserTest.GetSnapshotFromRendererTest
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