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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 <sstream> | 11 #include <sstream> |
12 #include <string> | 12 #include <string> |
13 | 13 |
14 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "testing/gtest/include/gtest/gtest.h" |
15 | 15 |
16 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
17 #include "webrtc/base/scoped_ptr.h" | 17 #include "webrtc/base/scoped_ptr.h" |
18 #include "webrtc/base/thread_annotations.h" | 18 #include "webrtc/base/thread_annotations.h" |
19 #include "webrtc/call.h" | 19 #include "webrtc/call.h" |
20 #include "webrtc/call/transport_adapter.h" | 20 #include "webrtc/call/transport_adapter.h" |
21 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" | 21 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" |
22 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" | 22 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
23 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 23 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
24 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 24 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
25 #include "webrtc/system_wrappers/include/rtp_to_ntp.h" | 25 #include "webrtc/system_wrappers/include/rtp_to_ntp.h" |
26 #include "webrtc/test/call_test.h" | 26 #include "webrtc/test/call_test.h" |
27 #include "webrtc/test/direct_transport.h" | 27 #include "webrtc/test/direct_transport.h" |
28 #include "webrtc/test/encoder_settings.h" | 28 #include "webrtc/test/encoder_settings.h" |
29 #include "webrtc/test/fake_audio_device.h" | 29 #include "webrtc/test/fake_audio_device.h" |
30 #include "webrtc/test/fake_decoder.h" | 30 #include "webrtc/test/fake_decoder.h" |
31 #include "webrtc/test/fake_encoder.h" | 31 #include "webrtc/test/fake_encoder.h" |
32 #include "webrtc/test/frame_generator.h" | 32 #include "webrtc/test/frame_generator.h" |
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706 int encoder_inits_; | 706 int encoder_inits_; |
707 uint32_t last_set_bitrate_; | 707 uint32_t last_set_bitrate_; |
708 VideoSendStream* send_stream_; | 708 VideoSendStream* send_stream_; |
709 VideoEncoderConfig encoder_config_; | 709 VideoEncoderConfig encoder_config_; |
710 } test; | 710 } test; |
711 | 711 |
712 RunBaseTest(&test, FakeNetworkPipe::Config()); | 712 RunBaseTest(&test, FakeNetworkPipe::Config()); |
713 } | 713 } |
714 | 714 |
715 } // namespace webrtc | 715 } // namespace webrtc |
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