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

Issue 1668493002: Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 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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803 803
804 WEBRTC_BASE_TEST(GetStats); 804 WEBRTC_BASE_TEST(GetStats);
805 WEBRTC_BASE_TEST(GetStatsMultipleRecvStreams); 805 WEBRTC_BASE_TEST(GetStatsMultipleRecvStreams);
806 WEBRTC_BASE_TEST(GetStatsMultipleSendStreams); 806 WEBRTC_BASE_TEST(GetStatsMultipleSendStreams);
807 807
808 WEBRTC_BASE_TEST(SetSendBandwidth); 808 WEBRTC_BASE_TEST(SetSendBandwidth);
809 809
810 WEBRTC_BASE_TEST(SetSendSsrc); 810 WEBRTC_BASE_TEST(SetSendSsrc);
811 WEBRTC_BASE_TEST(SetSendSsrcAfterSetCodecs); 811 WEBRTC_BASE_TEST(SetSendSsrcAfterSetCodecs);
812 812
813 WEBRTC_BASE_TEST(SetRenderer); 813 WEBRTC_BASE_TEST(SetSink);
814 814
815 WEBRTC_BASE_TEST(AddRemoveSendStreams); 815 WEBRTC_BASE_TEST(AddRemoveSendStreams);
816 816
817 WEBRTC_BASE_TEST(SimulateConference); 817 WEBRTC_BASE_TEST(SimulateConference);
818 818
819 WEBRTC_BASE_TEST(AddRemoveCapturer); 819 WEBRTC_BASE_TEST(AddRemoveCapturer);
820 820
821 WEBRTC_BASE_TEST(RemoveCapturerWithoutAdd); 821 WEBRTC_BASE_TEST(RemoveCapturerWithoutAdd);
822 822
823 WEBRTC_BASE_TEST(AddRemoveCapturerMultipleSources); 823 WEBRTC_BASE_TEST(AddRemoveCapturerMultipleSources);
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1811 1811
1812 TEST_F(WebRtcVideoChannel2Test, EstimatesNtpStartTimeCorrectly) { 1812 TEST_F(WebRtcVideoChannel2Test, EstimatesNtpStartTimeCorrectly) {
1813 // Start at last timestamp to verify that wraparounds are estimated correctly. 1813 // Start at last timestamp to verify that wraparounds are estimated correctly.
1814 static const uint32_t kInitialTimestamp = 0xFFFFFFFFu; 1814 static const uint32_t kInitialTimestamp = 0xFFFFFFFFu;
1815 static const int64_t kInitialNtpTimeMs = 1247891230; 1815 static const int64_t kInitialNtpTimeMs = 1247891230;
1816 static const int kFrameOffsetMs = 20; 1816 static const int kFrameOffsetMs = 20;
1817 EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); 1817 EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
1818 1818
1819 FakeVideoReceiveStream* stream = AddRecvStream(); 1819 FakeVideoReceiveStream* stream = AddRecvStream();
1820 cricket::FakeVideoRenderer renderer; 1820 cricket::FakeVideoRenderer renderer;
1821 EXPECT_TRUE(channel_->SetRenderer(last_ssrc_, &renderer)); 1821 EXPECT_TRUE(channel_->SetSink(last_ssrc_, &renderer));
1822 1822
1823 webrtc::VideoFrame video_frame; 1823 webrtc::VideoFrame video_frame;
1824 CreateBlackFrame(&video_frame, 4, 4); 1824 CreateBlackFrame(&video_frame, 4, 4);
1825 video_frame.set_timestamp(kInitialTimestamp); 1825 video_frame.set_timestamp(kInitialTimestamp);
1826 // Initial NTP time is not available on the first frame, but should still be 1826 // Initial NTP time is not available on the first frame, but should still be
1827 // able to be estimated. 1827 // able to be estimated.
1828 stream->InjectFrame(video_frame, 0); 1828 stream->InjectFrame(video_frame, 0);
1829 1829
1830 EXPECT_EQ(1, renderer.num_rendered_frames()); 1830 EXPECT_EQ(1, renderer.num_rendered_frames());
1831 1831
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3097 // Test that we normalize send codec format size in simulcast. 3097 // Test that we normalize send codec format size in simulcast.
3098 TEST_F(WebRtcVideoChannel2SimulcastTest, SetSendCodecsWithOddSizeInSimulcast) { 3098 TEST_F(WebRtcVideoChannel2SimulcastTest, SetSendCodecsWithOddSizeInSimulcast) {
3099 cricket::VideoCodec codec(kVp8Codec270p); 3099 cricket::VideoCodec codec(kVp8Codec270p);
3100 codec.width += 1; 3100 codec.width += 1;
3101 codec.height += 1; 3101 codec.height += 1;
3102 VerifySimulcastSettings(codec, 2, 2); 3102 VerifySimulcastSettings(codec, 2, 2);
3103 } 3103 }
3104 } // namespace cricket 3104 } // namespace cricket
3105 3105
3106 #endif // HAVE_WEBRTC_VIDEO 3106 #endif // HAVE_WEBRTC_VIDEO
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