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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 "webrtc/logging/rtc_event_log/rtc_event_log.h" | 11 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| 12 | 12 |
| 13 #include <limits> | 13 #include <limits> |
| 14 #include <vector> | 14 #include <vector> |
| 15 | 15 |
| 16 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
| 17 #include "webrtc/base/constructormagic.h" | 17 #include "webrtc/base/constructormagic.h" |
| 18 #include "webrtc/base/event.h" | 18 #include "webrtc/base/event.h" |
| 19 #include "webrtc/base/logging.h" | 19 #include "webrtc/base/logging.h" |
| 20 #include "webrtc/base/protobuf_utils.h" |
| 20 #include "webrtc/base/swap_queue.h" | 21 #include "webrtc/base/swap_queue.h" |
| 21 #include "webrtc/base/thread_checker.h" | 22 #include "webrtc/base/thread_checker.h" |
| 22 #include "webrtc/base/timeutils.h" | 23 #include "webrtc/base/timeutils.h" |
| 23 #include "webrtc/call/call.h" | 24 #include "webrtc/call/call.h" |
| 24 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h" | 25 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h" |
| 25 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_networ
k_adaptor.h" | 26 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_networ
k_adaptor.h" |
| 26 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 27 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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/rtp_rtcp/source/rtcp_packet/app.h" | 29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" |
| 29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" | 30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| 30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" | 31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
| 31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" | 32 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
| 32 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" | 33 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| 33 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h" | 34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h" |
| 34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" | 35 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| 35 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" | 36 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" |
| 36 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" | 37 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" |
| 37 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" | 38 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| 38 #include "webrtc/system_wrappers/include/file_wrapper.h" | 39 #include "webrtc/system_wrappers/include/file_wrapper.h" |
| 39 | 40 |
| 40 #ifdef ENABLE_RTC_EVENT_LOG | 41 #ifdef ENABLE_RTC_EVENT_LOG |
| 41 // Files generated at build-time by the protobuf compiler. | 42 // *.pb.h files are generated at build-time by the protobuf compiler. |
| 42 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | 43 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| 43 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h" | 44 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
| 44 #else | 45 #else |
| 45 #include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h" | 46 #include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
| 46 #endif | 47 #endif |
| 47 #endif | 48 #endif |
| 48 | 49 |
| 49 namespace webrtc { | 50 namespace webrtc { |
| 50 | 51 |
| 51 #ifdef ENABLE_RTC_EVENT_LOG | 52 #ifdef ENABLE_RTC_EVENT_LOG |
| (...skipping 525 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 577 } | 578 } |
| 578 | 579 |
| 579 bool RtcEventLog::ParseRtcEventLog(const std::string& file_name, | 580 bool RtcEventLog::ParseRtcEventLog(const std::string& file_name, |
| 580 rtclog::EventStream* result) { | 581 rtclog::EventStream* result) { |
| 581 char tmp_buffer[1024]; | 582 char tmp_buffer[1024]; |
| 582 int bytes_read = 0; | 583 int bytes_read = 0; |
| 583 std::unique_ptr<FileWrapper> dump_file(FileWrapper::Create()); | 584 std::unique_ptr<FileWrapper> dump_file(FileWrapper::Create()); |
| 584 if (!dump_file->OpenFile(file_name.c_str(), true)) { | 585 if (!dump_file->OpenFile(file_name.c_str(), true)) { |
| 585 return false; | 586 return false; |
| 586 } | 587 } |
| 587 std::string dump_buffer; | 588 ProtoString dump_buffer; |
| 588 while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) { | 589 while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) { |
| 589 dump_buffer.append(tmp_buffer, bytes_read); | 590 dump_buffer.append(tmp_buffer, bytes_read); |
| 590 } | 591 } |
| 591 dump_file->CloseFile(); | 592 dump_file->CloseFile(); |
| 592 return result->ParseFromString(dump_buffer); | 593 return result->ParseFromString(dump_buffer); |
| 593 } | 594 } |
| 594 | 595 |
| 595 #endif // ENABLE_RTC_EVENT_LOG | 596 #endif // ENABLE_RTC_EVENT_LOG |
| 596 | 597 |
| 597 bool RtcEventLogNullImpl::StartLogging(rtc::PlatformFile platform_file, | 598 bool RtcEventLogNullImpl::StartLogging(rtc::PlatformFile platform_file, |
| (...skipping 12 matching lines...) Expand all Loading... |
| 610 #else | 611 #else |
| 611 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); | 612 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); |
| 612 #endif // ENABLE_RTC_EVENT_LOG | 613 #endif // ENABLE_RTC_EVENT_LOG |
| 613 } | 614 } |
| 614 | 615 |
| 615 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() { | 616 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() { |
| 616 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); | 617 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); |
| 617 } | 618 } |
| 618 | 619 |
| 619 } // namespace webrtc | 620 } // namespace webrtc |
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