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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/swap_queue.h" | 19 #include "webrtc/base/swap_queue.h" |
20 #include "webrtc/base/thread_checker.h" | 20 #include "webrtc/base/thread_checker.h" |
21 #include "webrtc/base/timeutils.h" | 21 #include "webrtc/base/timeutils.h" |
22 #include "webrtc/call/call.h" | 22 #include "webrtc/call/call.h" |
23 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h" | 23 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h" |
| 24 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_networ
k_adaptor.h" |
24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
25 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 26 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
26 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" | 27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" |
27 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" | 28 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
28 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" | 29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
29 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" | 30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
30 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" | 31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| 32 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h" |
31 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" | 33 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
32 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h" | |
33 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" | 34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" |
34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" | 35 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" |
35 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" | 36 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
36 #include "webrtc/system_wrappers/include/file_wrapper.h" | 37 #include "webrtc/system_wrappers/include/file_wrapper.h" |
37 #include "webrtc/system_wrappers/include/logging.h" | 38 #include "webrtc/system_wrappers/include/logging.h" |
38 | 39 |
39 #ifdef ENABLE_RTC_EVENT_LOG | 40 #ifdef ENABLE_RTC_EVENT_LOG |
40 // Files generated at build-time by the protobuf compiler. | 41 // Files generated at build-time by the protobuf compiler. |
41 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | 42 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
42 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h" | 43 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
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78 MediaType media_type, | 79 MediaType media_type, |
79 const uint8_t* packet, | 80 const uint8_t* packet, |
80 size_t length) override; | 81 size_t length) override; |
81 void LogAudioPlayout(uint32_t ssrc) override; | 82 void LogAudioPlayout(uint32_t ssrc) override; |
82 void LogLossBasedBweUpdate(int32_t bitrate_bps, | 83 void LogLossBasedBweUpdate(int32_t bitrate_bps, |
83 uint8_t fraction_loss, | 84 uint8_t fraction_loss, |
84 int32_t total_packets) override; | 85 int32_t total_packets) override; |
85 void LogDelayBasedBweUpdate(int32_t bitrate_bps, | 86 void LogDelayBasedBweUpdate(int32_t bitrate_bps, |
86 BandwidthUsage detector_state) override; | 87 BandwidthUsage detector_state) override; |
87 void LogAudioNetworkAdaptation( | 88 void LogAudioNetworkAdaptation( |
88 const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override; | 89 const AudioEncoderRuntimeConfig& config) override; |
89 void LogProbeClusterCreated(int id, | 90 void LogProbeClusterCreated(int id, |
90 int bitrate_bps, | 91 int bitrate_bps, |
91 int min_probes, | 92 int min_probes, |
92 int min_bytes) override; | 93 int min_bytes) override; |
93 void LogProbeResultSuccess(int id, int bitrate_bps) override; | 94 void LogProbeResultSuccess(int id, int bitrate_bps) override; |
94 void LogProbeResultFailure(int id, | 95 void LogProbeResultFailure(int id, |
95 ProbeFailureReason failure_reason) override; | 96 ProbeFailureReason failure_reason) override; |
96 | 97 |
97 private: | 98 private: |
98 void StoreEvent(std::unique_ptr<rtclog::Event>* event); | 99 void StoreEvent(std::unique_ptr<rtclog::Event>* event); |
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497 std::unique_ptr<rtclog::Event> event(new rtclog::Event()); | 498 std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
498 event->set_timestamp_us(rtc::TimeMicros()); | 499 event->set_timestamp_us(rtc::TimeMicros()); |
499 event->set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE); | 500 event->set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE); |
500 auto bwe_event = event->mutable_delay_based_bwe_update(); | 501 auto bwe_event = event->mutable_delay_based_bwe_update(); |
501 bwe_event->set_bitrate_bps(bitrate_bps); | 502 bwe_event->set_bitrate_bps(bitrate_bps); |
502 bwe_event->set_detector_state(ConvertDetectorState(detector_state)); | 503 bwe_event->set_detector_state(ConvertDetectorState(detector_state)); |
503 StoreEvent(&event); | 504 StoreEvent(&event); |
504 } | 505 } |
505 | 506 |
506 void RtcEventLogImpl::LogAudioNetworkAdaptation( | 507 void RtcEventLogImpl::LogAudioNetworkAdaptation( |
507 const AudioNetworkAdaptor::EncoderRuntimeConfig& config) { | 508 const AudioEncoderRuntimeConfig& config) { |
508 std::unique_ptr<rtclog::Event> event(new rtclog::Event()); | 509 std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
509 event->set_timestamp_us(rtc::TimeMicros()); | 510 event->set_timestamp_us(rtc::TimeMicros()); |
510 event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT); | 511 event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT); |
511 auto audio_network_adaptation = event->mutable_audio_network_adaptation(); | 512 auto audio_network_adaptation = event->mutable_audio_network_adaptation(); |
512 if (config.bitrate_bps) | 513 if (config.bitrate_bps) |
513 audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps); | 514 audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps); |
514 if (config.frame_length_ms) | 515 if (config.frame_length_ms) |
515 audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms); | 516 audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms); |
516 if (config.uplink_packet_loss_fraction) { | 517 if (config.uplink_packet_loss_fraction) { |
517 audio_network_adaptation->set_uplink_packet_loss_fraction( | 518 audio_network_adaptation->set_uplink_packet_loss_fraction( |
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609 #else | 610 #else |
610 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); | 611 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); |
611 #endif // ENABLE_RTC_EVENT_LOG | 612 #endif // ENABLE_RTC_EVENT_LOG |
612 } | 613 } |
613 | 614 |
614 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() { | 615 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() { |
615 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); | 616 return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); |
616 } | 617 } |
617 | 618 |
618 } // namespace webrtc | 619 } // namespace webrtc |
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