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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> |
11 #include <map> | 11 #include <map> |
12 #include <sstream> | 12 #include <sstream> |
13 #include <string> | 13 #include <string> |
14 | 14 |
15 #include "testing/gtest/include/gtest/gtest.h" | 15 #include "testing/gtest/include/gtest/gtest.h" |
16 | 16 |
17 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
18 #include "webrtc/base/event.h" | 18 #include "webrtc/base/event.h" |
19 #include "webrtc/base/scoped_ptr.h" | 19 #include "webrtc/base/scoped_ptr.h" |
20 #include "webrtc/call.h" | 20 #include "webrtc/call.h" |
| 21 #include "webrtc/call/transport_adapter.h" |
21 #include "webrtc/frame_callback.h" | 22 #include "webrtc/frame_callback.h" |
22 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 23 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
23 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | 24 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" |
24 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" | 25 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
25 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" | 26 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" |
26 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 27 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
27 #include "webrtc/system_wrappers/interface/event_wrapper.h" | 28 #include "webrtc/system_wrappers/interface/event_wrapper.h" |
28 #include "webrtc/system_wrappers/interface/metrics.h" | 29 #include "webrtc/system_wrappers/interface/metrics.h" |
29 #include "webrtc/system_wrappers/interface/sleep.h" | 30 #include "webrtc/system_wrappers/interface/sleep.h" |
30 #include "webrtc/test/call_test.h" | 31 #include "webrtc/test/call_test.h" |
31 #include "webrtc/test/direct_transport.h" | 32 #include "webrtc/test/direct_transport.h" |
32 #include "webrtc/test/encoder_settings.h" | 33 #include "webrtc/test/encoder_settings.h" |
33 #include "webrtc/test/fake_audio_device.h" | 34 #include "webrtc/test/fake_audio_device.h" |
34 #include "webrtc/test/fake_decoder.h" | 35 #include "webrtc/test/fake_decoder.h" |
35 #include "webrtc/test/fake_encoder.h" | 36 #include "webrtc/test/fake_encoder.h" |
36 #include "webrtc/test/frame_generator.h" | 37 #include "webrtc/test/frame_generator.h" |
37 #include "webrtc/test/frame_generator_capturer.h" | 38 #include "webrtc/test/frame_generator_capturer.h" |
38 #include "webrtc/test/histogram.h" | 39 #include "webrtc/test/histogram.h" |
39 #include "webrtc/test/null_transport.h" | 40 #include "webrtc/test/null_transport.h" |
40 #include "webrtc/test/rtcp_packet_parser.h" | 41 #include "webrtc/test/rtcp_packet_parser.h" |
41 #include "webrtc/test/rtp_rtcp_observer.h" | 42 #include "webrtc/test/rtp_rtcp_observer.h" |
42 #include "webrtc/test/testsupport/fileutils.h" | 43 #include "webrtc/test/testsupport/fileutils.h" |
43 #include "webrtc/test/testsupport/gtest_disable.h" | 44 #include "webrtc/test/testsupport/gtest_disable.h" |
44 #include "webrtc/test/testsupport/perf_test.h" | 45 #include "webrtc/test/testsupport/perf_test.h" |
45 #include "webrtc/video/transport_adapter.h" | |
46 #include "webrtc/video_encoder.h" | 46 #include "webrtc/video_encoder.h" |
47 | 47 |
48 namespace webrtc { | 48 namespace webrtc { |
49 | 49 |
50 static const unsigned long kSilenceTimeoutMs = 2000; | 50 static const unsigned long kSilenceTimeoutMs = 2000; |
51 | 51 |
52 class EndToEndTest : public test::CallTest { | 52 class EndToEndTest : public test::CallTest { |
53 public: | 53 public: |
54 EndToEndTest() {} | 54 EndToEndTest() {} |
55 | 55 |
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3122 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) | 3122 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) |
3123 << "Enabling RTX requires rtpmap: rtx negotiation."; | 3123 << "Enabling RTX requires rtpmap: rtx negotiation."; |
3124 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 3124 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
3125 << "Enabling RTP extensions require negotiation."; | 3125 << "Enabling RTP extensions require negotiation."; |
3126 | 3126 |
3127 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 3127 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
3128 VerifyEmptyFecConfig(default_receive_config.rtp.fec); | 3128 VerifyEmptyFecConfig(default_receive_config.rtp.fec); |
3129 } | 3129 } |
3130 | 3130 |
3131 } // namespace webrtc | 3131 } // namespace webrtc |
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