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Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month ago
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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/call/transport_adapter.h"
22 #include "webrtc/frame_callback.h" 22 #include "webrtc/frame_callback.h"
23 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 23 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
25 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" 25 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
26 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" 26 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
27 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" 27 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
28 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 28 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
29 #include "webrtc/system_wrappers/interface/event_wrapper.h" 29 #include "webrtc/system_wrappers/include/event_wrapper.h"
30 #include "webrtc/system_wrappers/interface/metrics.h" 30 #include "webrtc/system_wrappers/include/metrics.h"
31 #include "webrtc/system_wrappers/interface/sleep.h" 31 #include "webrtc/system_wrappers/include/sleep.h"
32 #include "webrtc/test/call_test.h" 32 #include "webrtc/test/call_test.h"
33 #include "webrtc/test/direct_transport.h" 33 #include "webrtc/test/direct_transport.h"
34 #include "webrtc/test/encoder_settings.h" 34 #include "webrtc/test/encoder_settings.h"
35 #include "webrtc/test/fake_audio_device.h" 35 #include "webrtc/test/fake_audio_device.h"
36 #include "webrtc/test/fake_decoder.h" 36 #include "webrtc/test/fake_decoder.h"
37 #include "webrtc/test/fake_encoder.h" 37 #include "webrtc/test/fake_encoder.h"
38 #include "webrtc/test/frame_generator.h" 38 #include "webrtc/test/frame_generator.h"
39 #include "webrtc/test/frame_generator_capturer.h" 39 #include "webrtc/test/frame_generator_capturer.h"
40 #include "webrtc/test/histogram.h" 40 #include "webrtc/test/histogram.h"
41 #include "webrtc/test/null_transport.h" 41 #include "webrtc/test/null_transport.h"
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3162 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) 3162 EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
3163 << "Enabling RTX requires rtpmap: rtx negotiation."; 3163 << "Enabling RTX requires rtpmap: rtx negotiation.";
3164 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) 3164 EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
3165 << "Enabling RTP extensions require negotiation."; 3165 << "Enabling RTP extensions require negotiation.";
3166 3166
3167 VerifyEmptyNackConfig(default_receive_config.rtp.nack); 3167 VerifyEmptyNackConfig(default_receive_config.rtp.nack);
3168 VerifyEmptyFecConfig(default_receive_config.rtp.fec); 3168 VerifyEmptyFecConfig(default_receive_config.rtp.fec);
3169 } 3169 }
3170 3170
3171 } // namespace webrtc 3171 } // namespace webrtc
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