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Issue 1417683006: modules: more interface -> include renames (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix last incorrectly wrapped pragma message 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 <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"
(...skipping 673 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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