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Side by Side Diff: talk/media/webrtc/webrtcvideoengine2_unittest.cc

Issue 1602283003: Delete GetRenderer method, used only by the tests. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Deleted testcases which become useless without GetRenderer Created 4 years, 11 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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796 WEBRTC_BASE_TEST(GetStats); 796 WEBRTC_BASE_TEST(GetStats);
797 WEBRTC_BASE_TEST(GetStatsMultipleRecvStreams); 797 WEBRTC_BASE_TEST(GetStatsMultipleRecvStreams);
798 WEBRTC_BASE_TEST(GetStatsMultipleSendStreams); 798 WEBRTC_BASE_TEST(GetStatsMultipleSendStreams);
799 799
800 WEBRTC_BASE_TEST(SetSendBandwidth); 800 WEBRTC_BASE_TEST(SetSendBandwidth);
801 801
802 WEBRTC_BASE_TEST(SetSendSsrc); 802 WEBRTC_BASE_TEST(SetSendSsrc);
803 WEBRTC_BASE_TEST(SetSendSsrcAfterSetCodecs); 803 WEBRTC_BASE_TEST(SetSendSsrcAfterSetCodecs);
804 804
805 WEBRTC_BASE_TEST(SetRenderer); 805 WEBRTC_BASE_TEST(SetRenderer);
806 WEBRTC_BASE_TEST(AddRemoveRecvStreams);
807 806
808 WEBRTC_DISABLED_BASE_TEST(AddRemoveRecvStreamAndRender); 807 WEBRTC_DISABLED_BASE_TEST(AddRemoveRecvStreamAndRender);
809 808
810 WEBRTC_BASE_TEST(AddRemoveRecvStreamsNoConference);
811
812 WEBRTC_BASE_TEST(AddRemoveSendStreams); 809 WEBRTC_BASE_TEST(AddRemoveSendStreams);
813 810
814 WEBRTC_BASE_TEST(SimulateConference); 811 WEBRTC_BASE_TEST(SimulateConference);
815 812
816 WEBRTC_BASE_TEST(AddRemoveCapturer); 813 WEBRTC_BASE_TEST(AddRemoveCapturer);
817 814
818 WEBRTC_BASE_TEST(RemoveCapturerWithoutAdd); 815 WEBRTC_BASE_TEST(RemoveCapturerWithoutAdd);
819 816
820 WEBRTC_BASE_TEST(AddRemoveCapturerMultipleSources); 817 WEBRTC_BASE_TEST(AddRemoveCapturerMultipleSources);
821 818
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3400 // Ensures that the correct settings are applied to the codec when two temporal 3397 // Ensures that the correct settings are applied to the codec when two temporal
3401 // layer screencasting is enabled, and that the correct simulcast settings are 3398 // layer screencasting is enabled, and that the correct simulcast settings are
3402 // reapplied when disabling screencasting. 3399 // reapplied when disabling screencasting.
3403 TEST_F(WebRtcVideoChannel2SimulcastTest, 3400 TEST_F(WebRtcVideoChannel2SimulcastTest,
3404 DISABLED_TwoTemporalLayerScreencastSettings) { 3401 DISABLED_TwoTemporalLayerScreencastSettings) {
3405 // TODO(pbos): Implement. 3402 // TODO(pbos): Implement.
3406 FAIL() << "Not implemented."; 3403 FAIL() << "Not implemented.";
3407 } 3404 }
3408 3405
3409 } // namespace cricket 3406 } // namespace cricket
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