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Issue 2767123002: Disable the ORTC integration tests on TSan. (Closed)
Patch Set: Created 3 years, 9 months ago
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1 /* 1 /*
2 * Copyright 2017 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2017 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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339 std::unique_ptr<OrtcFactoryInterface> ortc_factory1_; 339 std::unique_ptr<OrtcFactoryInterface> ortc_factory1_;
340 std::unique_ptr<OrtcFactoryInterface> ortc_factory2_; 340 std::unique_ptr<OrtcFactoryInterface> ortc_factory2_;
341 // Actually owned by video tracks. 341 // Actually owned by video tracks.
342 std::vector<cricket::FakeVideoCapturer*> fake_video_capturers_; 342 std::vector<cricket::FakeVideoCapturer*> fake_video_capturers_;
343 int received_audio_frames1_ = 0; 343 int received_audio_frames1_ = 0;
344 int received_audio_frames2_ = 0; 344 int received_audio_frames2_ = 0;
345 int rendered_video_frames1_ = 0; 345 int rendered_video_frames1_ = 0;
346 int rendered_video_frames2_ = 0; 346 int rendered_video_frames2_ = 0;
347 }; 347 };
348 348
349 // Disable for TSan v2, see
350 // https://bugs.chromium.org/p/webrtc/issues/detail?id=7366 for details.
351 #if !defined(THREAD_SANITIZER)
352
349 // Very basic end-to-end test with a single pair of audio RTP sender and 353 // Very basic end-to-end test with a single pair of audio RTP sender and
350 // receiver. 354 // receiver.
351 // 355 //
352 // Uses muxed RTCP, and minimal parameters with a hard-coded config that's 356 // Uses muxed RTCP, and minimal parameters with a hard-coded config that's
353 // known to work. 357 // known to work.
354 TEST_F(OrtcFactoryIntegrationTest, BasicOneWayAudioRtpSenderAndReceiver) { 358 TEST_F(OrtcFactoryIntegrationTest, BasicOneWayAudioRtpSenderAndReceiver) {
355 auto udp_transports = CreateAndConnectUdpTransportPair(); 359 auto udp_transports = CreateAndConnectUdpTransportPair();
356 auto rtp_transports = 360 auto rtp_transports =
357 CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports); 361 CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
358 362
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668 672
669 // TODO(deadbeef): End-to-end test for multiple senders/receivers of the same 673 // TODO(deadbeef): End-to-end test for multiple senders/receivers of the same
670 // media type, once that's supported. Currently, it is not because the 674 // media type, once that's supported. Currently, it is not because the
671 // BaseChannel model relies on there being a single VoiceChannel and 675 // BaseChannel model relies on there being a single VoiceChannel and
672 // VideoChannel, and these only support a single set of codecs/etc. per 676 // VideoChannel, and these only support a single set of codecs/etc. per
673 // send/receive direction. 677 // send/receive direction.
674 678
675 // TODO(deadbeef): End-to-end test for simulcast, once that's supported by this 679 // TODO(deadbeef): End-to-end test for simulcast, once that's supported by this
676 // API. 680 // API.
677 681
682 #endif // if !defined(THREAD_SANITIZER)
683
678 } // namespace webrtc 684 } // namespace webrtc
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