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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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