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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #include <algorithm> | 10 #include <algorithm> |
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34 #include "webrtc/test/fake_audio_device.h" | 34 #include "webrtc/test/fake_audio_device.h" |
35 #include "webrtc/test/fake_decoder.h" | 35 #include "webrtc/test/fake_decoder.h" |
36 #include "webrtc/test/fake_encoder.h" | 36 #include "webrtc/test/fake_encoder.h" |
37 #include "webrtc/test/frame_generator.h" | 37 #include "webrtc/test/frame_generator.h" |
38 #include "webrtc/test/frame_generator_capturer.h" | 38 #include "webrtc/test/frame_generator_capturer.h" |
39 #include "webrtc/test/histogram.h" | 39 #include "webrtc/test/histogram.h" |
40 #include "webrtc/test/null_transport.h" | 40 #include "webrtc/test/null_transport.h" |
41 #include "webrtc/test/rtcp_packet_parser.h" | 41 #include "webrtc/test/rtcp_packet_parser.h" |
42 #include "webrtc/test/rtp_rtcp_observer.h" | 42 #include "webrtc/test/rtp_rtcp_observer.h" |
43 #include "webrtc/test/testsupport/fileutils.h" | 43 #include "webrtc/test/testsupport/fileutils.h" |
44 #include "webrtc/test/testsupport/gtest_disable.h" | |
45 #include "webrtc/test/testsupport/perf_test.h" | 44 #include "webrtc/test/testsupport/perf_test.h" |
46 #include "webrtc/video_encoder.h" | 45 #include "webrtc/video_encoder.h" |
47 | 46 |
48 namespace webrtc { | 47 namespace webrtc { |
49 | 48 |
50 static const int kSilenceTimeoutMs = 2000; | 49 static const int kSilenceTimeoutMs = 2000; |
51 | 50 |
52 class EndToEndTest : public test::CallTest { | 51 class EndToEndTest : public test::CallTest { |
53 public: | 52 public: |
54 EndToEndTest() {} | 53 EndToEndTest() {} |
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3244 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) | 3243 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) |
3245 << "Enabling RTX requires rtpmap: rtx negotiation."; | 3244 << "Enabling RTX requires rtpmap: rtx negotiation."; |
3246 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 3245 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
3247 << "Enabling RTP extensions require negotiation."; | 3246 << "Enabling RTP extensions require negotiation."; |
3248 | 3247 |
3249 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 3248 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
3250 VerifyEmptyFecConfig(default_receive_config.rtp.fec); | 3249 VerifyEmptyFecConfig(default_receive_config.rtp.fec); |
3251 } | 3250 } |
3252 | 3251 |
3253 } // namespace webrtc | 3252 } // namespace webrtc |
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