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Issue 1417773002: Remove system_wrappers/interface/trace_event.h (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months 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 10
11 #include <string.h> 11 #include <string.h>
12 12
13 #include <map> 13 #include <map>
14 #include <vector> 14 #include <vector>
15 15
16 #include "webrtc/audio/audio_receive_stream.h" 16 #include "webrtc/audio/audio_receive_stream.h"
17 #include "webrtc/audio/audio_send_stream.h" 17 #include "webrtc/audio/audio_send_stream.h"
18 #include "webrtc/base/checks.h" 18 #include "webrtc/base/checks.h"
19 #include "webrtc/base/scoped_ptr.h" 19 #include "webrtc/base/scoped_ptr.h"
20 #include "webrtc/base/thread_annotations.h" 20 #include "webrtc/base/thread_annotations.h"
21 #include "webrtc/base/thread_checker.h" 21 #include "webrtc/base/thread_checker.h"
22 #include "webrtc/base/trace_event.h"
22 #include "webrtc/call.h" 23 #include "webrtc/call.h"
23 #include "webrtc/call/rtc_event_log.h" 24 #include "webrtc/call/rtc_event_log.h"
24 #include "webrtc/common.h" 25 #include "webrtc/common.h"
25 #include "webrtc/config.h" 26 #include "webrtc/config.h"
26 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 27 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
27 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 28 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
28 #include "webrtc/modules/utility/interface/process_thread.h" 29 #include "webrtc/modules/utility/interface/process_thread.h"
29 #include "webrtc/system_wrappers/interface/cpu_info.h" 30 #include "webrtc/system_wrappers/interface/cpu_info.h"
30 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 31 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
31 #include "webrtc/system_wrappers/interface/logging.h" 32 #include "webrtc/system_wrappers/interface/logging.h"
32 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" 33 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
33 #include "webrtc/system_wrappers/interface/trace.h" 34 #include "webrtc/system_wrappers/interface/trace.h"
34 #include "webrtc/system_wrappers/interface/trace_event.h"
35 #include "webrtc/video/video_receive_stream.h" 35 #include "webrtc/video/video_receive_stream.h"
36 #include "webrtc/video/video_send_stream.h" 36 #include "webrtc/video/video_send_stream.h"
37 #include "webrtc/voice_engine/include/voe_codec.h" 37 #include "webrtc/voice_engine/include/voe_codec.h"
38 38
39 namespace webrtc { 39 namespace webrtc {
40 40
41 const int Call::Config::kDefaultStartBitrateBps = 300000; 41 const int Call::Config::kDefaultStartBitrateBps = 300000;
42 42
43 namespace internal { 43 namespace internal {
44 44
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592 // thread. Then this check can be enabled. 592 // thread. Then this check can be enabled.
593 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); 593 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
594 if (RtpHeaderParser::IsRtcp(packet, length)) 594 if (RtpHeaderParser::IsRtcp(packet, length))
595 return DeliverRtcp(media_type, packet, length); 595 return DeliverRtcp(media_type, packet, length);
596 596
597 return DeliverRtp(media_type, packet, length, packet_time); 597 return DeliverRtp(media_type, packet, length, packet_time);
598 } 598 }
599 599
600 } // namespace internal 600 } // namespace internal
601 } // namespace webrtc 601 } // namespace webrtc
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