Original issue's description:
> Revert of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #18 id:340001 of https://codereview.webrtc.org/3007473002/ )
>
> Reason for revert:
> Breaks google3 project.
>
> Original issue's description:
> > Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread
> >
> > Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread. This will eventually allow us to run multiple log sessions on a single task-queue.
> >
> > BUG=webrtc:8142, webrtc:8143, webrtc:8145
> >
> > Review-Url: https://codereview.webrtc.org/3007473002
> > Cr-Commit-Position: refs/heads/master@{#19666}
> > Committed: f33cee7534
TBR=terelius@webrtc.org,nisse@webrtc.org,charujain@webrtc.org
NOPRESUBMIT=true
NOTRY=True
BUG=webrtc:8142, webrtc:8143, webrtc:8145
Review-Url: https://codereview.webrtc.org/3012783002
Cr-Commit-Position: refs/heads/master@{#19712}
849 lines
30 KiB
C++
849 lines
30 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
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#include <atomic>
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#include <deque>
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#include <functional>
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#include <limits>
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#include <memory>
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#include <utility>
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#include <vector>
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#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/event.h"
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#include "webrtc/rtc_base/ignore_wundef.h"
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#include "webrtc/rtc_base/logging.h"
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#include "webrtc/rtc_base/protobuf_utils.h"
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#include "webrtc/rtc_base/ptr_util.h"
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#include "webrtc/rtc_base/sequenced_task_checker.h"
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#include "webrtc/rtc_base/task_queue.h"
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#include "webrtc/rtc_base/thread_annotations.h"
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#include "webrtc/rtc_base/timeutils.h"
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#include "webrtc/system_wrappers/include/file_wrapper.h"
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#include "webrtc/typedefs.h"
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#ifdef ENABLE_RTC_EVENT_LOG
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// *.pb.h files are generated at build-time by the protobuf compiler.
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RTC_PUSH_IGNORING_WUNDEF()
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
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#else
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#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
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#endif
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RTC_POP_IGNORING_WUNDEF()
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#endif
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namespace webrtc {
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#ifdef ENABLE_RTC_EVENT_LOG
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namespace {
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const int kEventsInHistory = 10000;
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bool IsConfigEvent(const rtclog::Event& event) {
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rtclog::Event_EventType event_type = event.type();
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return event_type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT ||
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event_type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT ||
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event_type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT ||
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event_type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT;
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}
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// TODO(eladalon): This class exists because C++11 doesn't allow transferring a
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// unique_ptr to a lambda (a copy constructor is required). We should get
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// rid of this when we move to C++14.
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template <typename T>
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class ResourceOwningTask final : public rtc::QueuedTask {
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public:
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ResourceOwningTask(std::unique_ptr<T> resource,
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std::function<void(std::unique_ptr<T>)> handler)
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: resource_(std::move(resource)), handler_(handler) {}
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bool Run() override {
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handler_(std::move(resource_));
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return true;
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}
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private:
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std::unique_ptr<T> resource_;
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std::function<void(std::unique_ptr<T>)> handler_;
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};
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} // namespace
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class RtcEventLogImpl final : public RtcEventLog {
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friend std::unique_ptr<RtcEventLog> RtcEventLog::Create();
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public:
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~RtcEventLogImpl() override;
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bool StartLogging(const std::string& file_name,
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int64_t max_size_bytes) override;
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bool StartLogging(rtc::PlatformFile platform_file,
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int64_t max_size_bytes) override;
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void StopLogging() override;
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void LogVideoReceiveStreamConfig(const rtclog::StreamConfig& config) override;
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void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override;
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void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
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void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override;
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void LogRtpHeader(PacketDirection direction,
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const uint8_t* header,
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size_t packet_length) override;
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void LogRtpHeader(PacketDirection direction,
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const uint8_t* header,
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size_t packet_length,
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int probe_cluster_id) override;
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void LogRtcpPacket(PacketDirection direction,
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const uint8_t* packet,
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size_t length) override;
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void LogAudioPlayout(uint32_t ssrc) override;
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void LogLossBasedBweUpdate(int32_t bitrate_bps,
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uint8_t fraction_loss,
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int32_t total_packets) override;
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void LogDelayBasedBweUpdate(int32_t bitrate_bps,
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BandwidthUsage detector_state) override;
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void LogAudioNetworkAdaptation(
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const AudioEncoderRuntimeConfig& config) override;
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void LogProbeClusterCreated(int id,
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int bitrate_bps,
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int min_probes,
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int min_bytes) override;
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void LogProbeResultSuccess(int id, int bitrate_bps) override;
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void LogProbeResultFailure(int id,
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ProbeFailureReason failure_reason) override;
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private:
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void StartLoggingInternal(std::unique_ptr<FileWrapper> file,
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int64_t max_size_bytes);
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RtcEventLogImpl(); // Creation is done by RtcEventLog::Create.
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void StoreEvent(std::unique_ptr<rtclog::Event> event);
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void LogProbeResult(int id,
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rtclog::BweProbeResult::ResultType result,
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int bitrate_bps);
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// Appends an event to the output protobuf string, returning true on success.
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// Fails and returns false in case the limit on output size prevents the
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// event from being added; in this case, the output string is left unchanged.
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bool AppendEventToString(rtclog::Event* event,
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ProtoString* output_string) RTC_WARN_UNUSED_RESULT;
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void LogToMemory(std::unique_ptr<rtclog::Event> event);
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void StartLogFile();
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void LogToFile(std::unique_ptr<rtclog::Event> event);
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void StopLogFile(int64_t stop_time);
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// Observe a limit on the number of concurrent logs, so as not to run into
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// OS-imposed limits on open files and/or threads/task-queues.
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// TODO(eladalon): Known issue - there's a race over |log_count_|.
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static std::atomic<int> log_count_;
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// Make sure that the event log is "managed" - created/destroyed, as well
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// as started/stopped - from the same thread/task-queue.
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rtc::SequencedTaskChecker owner_sequence_checker_;
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// History containing all past configuration events.
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std::vector<std::unique_ptr<rtclog::Event>> config_history_
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ACCESS_ON(task_queue_);
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// History containing the most recent (non-configuration) events (~10s).
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std::deque<std::unique_ptr<rtclog::Event>> history_ ACCESS_ON(task_queue_);
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std::unique_ptr<FileWrapper> file_ ACCESS_ON(task_queue_);
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size_t max_size_bytes_ ACCESS_ON(task_queue_);
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size_t written_bytes_ ACCESS_ON(task_queue_);
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// Keep this last to ensure it destructs first, or else tasks living on the
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// queue might access other members after they've been torn down.
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rtc::TaskQueue task_queue_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogImpl);
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};
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namespace {
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// The functions in this namespace convert enums from the runtime format
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// that the rest of the WebRtc project can use, to the corresponding
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// serialized enum which is defined by the protobuf.
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rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
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switch (rtcp_mode) {
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case RtcpMode::kCompound:
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return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
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case RtcpMode::kReducedSize:
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return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
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case RtcpMode::kOff:
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RTC_NOTREACHED();
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return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
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}
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RTC_NOTREACHED();
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return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
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}
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rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState(
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BandwidthUsage state) {
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switch (state) {
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case BandwidthUsage::kBwNormal:
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return rtclog::DelayBasedBweUpdate::BWE_NORMAL;
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case BandwidthUsage::kBwUnderusing:
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return rtclog::DelayBasedBweUpdate::BWE_UNDERUSING;
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case BandwidthUsage::kBwOverusing:
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return rtclog::DelayBasedBweUpdate::BWE_OVERUSING;
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}
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RTC_NOTREACHED();
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return rtclog::DelayBasedBweUpdate::BWE_NORMAL;
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}
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rtclog::BweProbeResult::ResultType ConvertProbeResultType(
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ProbeFailureReason failure_reason) {
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switch (failure_reason) {
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case kInvalidSendReceiveInterval:
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return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL;
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case kInvalidSendReceiveRatio:
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return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO;
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case kTimeout:
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return rtclog::BweProbeResult::TIMEOUT;
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}
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RTC_NOTREACHED();
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return rtclog::BweProbeResult::SUCCESS;
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}
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} // namespace
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std::atomic<int> RtcEventLogImpl::log_count_(0);
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RtcEventLogImpl::RtcEventLogImpl()
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: file_(FileWrapper::Create()),
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max_size_bytes_(std::numeric_limits<decltype(max_size_bytes_)>::max()),
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written_bytes_(0),
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task_queue_("rtc_event_log") {}
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RtcEventLogImpl::~RtcEventLogImpl() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
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// If we're logging to the file, this will stop that. Blocking function.
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StopLogging();
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int count = std::atomic_fetch_sub(&RtcEventLogImpl::log_count_, 1) - 1;
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RTC_DCHECK_GE(count, 0);
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}
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bool RtcEventLogImpl::StartLogging(const std::string& file_name,
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int64_t max_size_bytes) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
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auto file = rtc::WrapUnique<FileWrapper>(FileWrapper::Create());
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if (!file->OpenFile(file_name.c_str(), false)) {
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LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
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return false;
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}
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StartLoggingInternal(std::move(file), max_size_bytes);
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return true;
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}
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bool RtcEventLogImpl::StartLogging(rtc::PlatformFile platform_file,
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int64_t max_size_bytes) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
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auto file = rtc::WrapUnique<FileWrapper>(FileWrapper::Create());
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FILE* file_handle = rtc::FdopenPlatformFileForWriting(platform_file);
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if (!file_handle) {
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LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
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// Even though we failed to open a FILE*, the platform_file is still open
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// and needs to be closed.
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if (!rtc::ClosePlatformFile(platform_file)) {
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LOG(LS_ERROR) << "Can't close file.";
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}
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return false;
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}
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if (!file->OpenFromFileHandle(file_handle)) {
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LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
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return false;
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}
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StartLoggingInternal(std::move(file), max_size_bytes);
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return true;
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}
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void RtcEventLogImpl::StopLogging() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
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LOG(LS_INFO) << "Stopping WebRTC event log.";
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const int64_t stop_time = rtc::TimeMicros();
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rtc::Event file_finished(true, false);
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task_queue_.PostTask([this, stop_time, &file_finished]() {
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RTC_DCHECK_RUN_ON(&task_queue_);
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if (file_->is_open()) {
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StopLogFile(stop_time);
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}
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file_finished.Set();
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});
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file_finished.Wait(rtc::Event::kForever);
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LOG(LS_INFO) << "WebRTC event log successfully stopped.";
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}
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void RtcEventLogImpl::LogVideoReceiveStreamConfig(
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const rtclog::StreamConfig& config) {
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std::unique_ptr<rtclog::Event> event(new rtclog::Event());
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event->set_timestamp_us(rtc::TimeMicros());
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event->set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
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rtclog::VideoReceiveConfig* receiver_config =
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event->mutable_video_receiver_config();
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receiver_config->set_remote_ssrc(config.remote_ssrc);
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receiver_config->set_local_ssrc(config.local_ssrc);
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// TODO(perkj): Add field for rsid.
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receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtcp_mode));
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receiver_config->set_remb(config.remb);
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for (const auto& e : config.rtp_extensions) {
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rtclog::RtpHeaderExtension* extension =
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receiver_config->add_header_extensions();
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extension->set_name(e.uri);
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extension->set_id(e.id);
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}
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for (const auto& d : config.codecs) {
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rtclog::DecoderConfig* decoder = receiver_config->add_decoders();
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decoder->set_name(d.payload_name);
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decoder->set_payload_type(d.payload_type);
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if (d.rtx_payload_type != 0) {
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rtclog::RtxMap* rtx = receiver_config->add_rtx_map();
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rtx->set_payload_type(d.payload_type);
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rtx->mutable_config()->set_rtx_ssrc(config.rtx_ssrc);
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rtx->mutable_config()->set_rtx_payload_type(d.rtx_payload_type);
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}
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}
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StoreEvent(std::move(event));
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}
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void RtcEventLogImpl::LogVideoSendStreamConfig(
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const rtclog::StreamConfig& config) {
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std::unique_ptr<rtclog::Event> event(new rtclog::Event());
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event->set_timestamp_us(rtc::TimeMicros());
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event->set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
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rtclog::VideoSendConfig* sender_config = event->mutable_video_sender_config();
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// TODO(perkj): rtclog::VideoSendConfig should only contain one SSRC.
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sender_config->add_ssrcs(config.local_ssrc);
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if (config.rtx_ssrc != 0) {
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sender_config->add_rtx_ssrcs(config.rtx_ssrc);
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}
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for (const auto& e : config.rtp_extensions) {
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rtclog::RtpHeaderExtension* extension =
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sender_config->add_header_extensions();
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extension->set_name(e.uri);
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extension->set_id(e.id);
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}
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// TODO(perkj): rtclog::VideoSendConfig should contain many possible codec
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// configurations.
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for (const auto& codec : config.codecs) {
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sender_config->set_rtx_payload_type(codec.rtx_payload_type);
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rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
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encoder->set_name(codec.payload_name);
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encoder->set_payload_type(codec.payload_type);
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if (config.codecs.size() > 1) {
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LOG(WARNING) << "LogVideoSendStreamConfig currently only supports one "
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<< "codec. Logging codec :" << codec.payload_name;
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break;
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}
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}
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StoreEvent(std::move(event));
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}
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void RtcEventLogImpl::LogAudioReceiveStreamConfig(
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const rtclog::StreamConfig& config) {
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std::unique_ptr<rtclog::Event> event(new rtclog::Event());
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event->set_timestamp_us(rtc::TimeMicros());
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event->set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
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rtclog::AudioReceiveConfig* receiver_config =
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event->mutable_audio_receiver_config();
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receiver_config->set_remote_ssrc(config.remote_ssrc);
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receiver_config->set_local_ssrc(config.local_ssrc);
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for (const auto& e : config.rtp_extensions) {
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rtclog::RtpHeaderExtension* extension =
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receiver_config->add_header_extensions();
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extension->set_name(e.uri);
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extension->set_id(e.id);
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}
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StoreEvent(std::move(event));
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}
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void RtcEventLogImpl::LogAudioSendStreamConfig(
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const rtclog::StreamConfig& config) {
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std::unique_ptr<rtclog::Event> event(new rtclog::Event());
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event->set_timestamp_us(rtc::TimeMicros());
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event->set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
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rtclog::AudioSendConfig* sender_config = event->mutable_audio_sender_config();
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sender_config->set_ssrc(config.local_ssrc);
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for (const auto& e : config.rtp_extensions) {
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rtclog::RtpHeaderExtension* extension =
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sender_config->add_header_extensions();
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extension->set_name(e.uri);
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extension->set_id(e.id);
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}
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StoreEvent(std::move(event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
|
|
const uint8_t* header,
|
|
size_t packet_length) {
|
|
LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
|
|
}
|
|
|
|
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
|
|
const uint8_t* header,
|
|
size_t packet_length,
|
|
int probe_cluster_id) {
|
|
// Read header length (in bytes) from packet data.
|
|
if (packet_length < 12u) {
|
|
return; // Don't read outside the packet.
|
|
}
|
|
const bool x = (header[0] & 0x10) != 0;
|
|
const uint8_t cc = header[0] & 0x0f;
|
|
size_t header_length = 12u + cc * 4u;
|
|
|
|
if (x) {
|
|
if (packet_length < 12u + cc * 4u + 4u) {
|
|
return; // Don't read outside the packet.
|
|
}
|
|
size_t x_len = ByteReader<uint16_t>::ReadBigEndian(header + 14 + cc * 4);
|
|
header_length += (x_len + 1) * 4;
|
|
}
|
|
|
|
std::unique_ptr<rtclog::Event> rtp_event(new rtclog::Event());
|
|
rtp_event->set_timestamp_us(rtc::TimeMicros());
|
|
rtp_event->set_type(rtclog::Event::RTP_EVENT);
|
|
rtp_event->mutable_rtp_packet()->set_incoming(direction == kIncomingPacket);
|
|
rtp_event->mutable_rtp_packet()->set_packet_length(packet_length);
|
|
rtp_event->mutable_rtp_packet()->set_header(header, header_length);
|
|
if (probe_cluster_id != PacedPacketInfo::kNotAProbe)
|
|
rtp_event->mutable_rtp_packet()->set_probe_cluster_id(probe_cluster_id);
|
|
StoreEvent(std::move(rtp_event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
|
|
const uint8_t* packet,
|
|
size_t length) {
|
|
std::unique_ptr<rtclog::Event> rtcp_event(new rtclog::Event());
|
|
rtcp_event->set_timestamp_us(rtc::TimeMicros());
|
|
rtcp_event->set_type(rtclog::Event::RTCP_EVENT);
|
|
rtcp_event->mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket);
|
|
|
|
rtcp::CommonHeader header;
|
|
const uint8_t* block_begin = packet;
|
|
const uint8_t* packet_end = packet + length;
|
|
RTC_DCHECK(length <= IP_PACKET_SIZE);
|
|
uint8_t buffer[IP_PACKET_SIZE];
|
|
uint32_t buffer_length = 0;
|
|
while (block_begin < packet_end) {
|
|
if (!header.Parse(block_begin, packet_end - block_begin)) {
|
|
break; // Incorrect message header.
|
|
}
|
|
const uint8_t* next_block = header.NextPacket();
|
|
uint32_t block_size = next_block - block_begin;
|
|
switch (header.type()) {
|
|
case rtcp::SenderReport::kPacketType:
|
|
case rtcp::ReceiverReport::kPacketType:
|
|
case rtcp::Bye::kPacketType:
|
|
case rtcp::ExtendedJitterReport::kPacketType:
|
|
case rtcp::Rtpfb::kPacketType:
|
|
case rtcp::Psfb::kPacketType:
|
|
case rtcp::ExtendedReports::kPacketType:
|
|
// We log sender reports, receiver reports, bye messages
|
|
// inter-arrival jitter, third-party loss reports, payload-specific
|
|
// feedback and extended reports.
|
|
memcpy(buffer + buffer_length, block_begin, block_size);
|
|
buffer_length += block_size;
|
|
break;
|
|
case rtcp::Sdes::kPacketType:
|
|
case rtcp::App::kPacketType:
|
|
default:
|
|
// We don't log sender descriptions, application defined messages
|
|
// or message blocks of unknown type.
|
|
break;
|
|
}
|
|
|
|
block_begin += block_size;
|
|
}
|
|
rtcp_event->mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
|
|
StoreEvent(std::move(rtcp_event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
|
|
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
|
|
event->set_timestamp_us(rtc::TimeMicros());
|
|
event->set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
|
|
auto playout_event = event->mutable_audio_playout_event();
|
|
playout_event->set_local_ssrc(ssrc);
|
|
StoreEvent(std::move(event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogLossBasedBweUpdate(int32_t bitrate_bps,
|
|
uint8_t fraction_loss,
|
|
int32_t total_packets) {
|
|
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
|
|
event->set_timestamp_us(rtc::TimeMicros());
|
|
event->set_type(rtclog::Event::LOSS_BASED_BWE_UPDATE);
|
|
auto bwe_event = event->mutable_loss_based_bwe_update();
|
|
bwe_event->set_bitrate_bps(bitrate_bps);
|
|
bwe_event->set_fraction_loss(fraction_loss);
|
|
bwe_event->set_total_packets(total_packets);
|
|
StoreEvent(std::move(event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogDelayBasedBweUpdate(int32_t bitrate_bps,
|
|
BandwidthUsage detector_state) {
|
|
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
|
|
event->set_timestamp_us(rtc::TimeMicros());
|
|
event->set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE);
|
|
auto bwe_event = event->mutable_delay_based_bwe_update();
|
|
bwe_event->set_bitrate_bps(bitrate_bps);
|
|
bwe_event->set_detector_state(ConvertDetectorState(detector_state));
|
|
StoreEvent(std::move(event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogAudioNetworkAdaptation(
|
|
const AudioEncoderRuntimeConfig& config) {
|
|
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
|
|
event->set_timestamp_us(rtc::TimeMicros());
|
|
event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
|
|
auto audio_network_adaptation = event->mutable_audio_network_adaptation();
|
|
if (config.bitrate_bps)
|
|
audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps);
|
|
if (config.frame_length_ms)
|
|
audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms);
|
|
if (config.uplink_packet_loss_fraction) {
|
|
audio_network_adaptation->set_uplink_packet_loss_fraction(
|
|
*config.uplink_packet_loss_fraction);
|
|
}
|
|
if (config.enable_fec)
|
|
audio_network_adaptation->set_enable_fec(*config.enable_fec);
|
|
if (config.enable_dtx)
|
|
audio_network_adaptation->set_enable_dtx(*config.enable_dtx);
|
|
if (config.num_channels)
|
|
audio_network_adaptation->set_num_channels(*config.num_channels);
|
|
StoreEvent(std::move(event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogProbeClusterCreated(int id,
|
|
int bitrate_bps,
|
|
int min_probes,
|
|
int min_bytes) {
|
|
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
|
|
event->set_timestamp_us(rtc::TimeMicros());
|
|
event->set_type(rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
|
|
|
|
auto probe_cluster = event->mutable_probe_cluster();
|
|
probe_cluster->set_id(id);
|
|
probe_cluster->set_bitrate_bps(bitrate_bps);
|
|
probe_cluster->set_min_packets(min_probes);
|
|
probe_cluster->set_min_bytes(min_bytes);
|
|
StoreEvent(std::move(event));
|
|
}
|
|
|
|
void RtcEventLogImpl::LogProbeResultSuccess(int id, int bitrate_bps) {
|
|
LogProbeResult(id, rtclog::BweProbeResult::SUCCESS, bitrate_bps);
|
|
}
|
|
|
|
void RtcEventLogImpl::LogProbeResultFailure(int id,
|
|
ProbeFailureReason failure_reason) {
|
|
rtclog::BweProbeResult::ResultType result =
|
|
ConvertProbeResultType(failure_reason);
|
|
LogProbeResult(id, result, -1);
|
|
}
|
|
|
|
void RtcEventLogImpl::LogProbeResult(int id,
|
|
rtclog::BweProbeResult::ResultType result,
|
|
int bitrate_bps) {
|
|
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
|
|
event->set_timestamp_us(rtc::TimeMicros());
|
|
event->set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
|
|
|
|
auto probe_result = event->mutable_probe_result();
|
|
probe_result->set_id(id);
|
|
probe_result->set_result(result);
|
|
if (result == rtclog::BweProbeResult::SUCCESS)
|
|
probe_result->set_bitrate_bps(bitrate_bps);
|
|
StoreEvent(std::move(event));
|
|
}
|
|
|
|
void RtcEventLogImpl::StartLoggingInternal(std::unique_ptr<FileWrapper> file,
|
|
int64_t max_size_bytes) {
|
|
LOG(LS_INFO) << "Starting WebRTC event log.";
|
|
|
|
max_size_bytes = (max_size_bytes <= 0)
|
|
? std::numeric_limits<decltype(max_size_bytes)>::max()
|
|
: max_size_bytes;
|
|
auto file_handler = [this,
|
|
max_size_bytes](std::unique_ptr<FileWrapper> file) {
|
|
RTC_DCHECK_RUN_ON(&task_queue_);
|
|
if (!file_->is_open()) {
|
|
max_size_bytes_ = max_size_bytes;
|
|
file_ = std::move(file);
|
|
StartLogFile();
|
|
} else {
|
|
// Already started. Ignore message and close file handle.
|
|
file->CloseFile();
|
|
}
|
|
};
|
|
task_queue_.PostTask(rtc::MakeUnique<ResourceOwningTask<FileWrapper>>(
|
|
std::move(file), file_handler));
|
|
}
|
|
|
|
void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event> event) {
|
|
RTC_DCHECK(event);
|
|
|
|
auto event_handler = [this](std::unique_ptr<rtclog::Event> rtclog_event) {
|
|
RTC_DCHECK_RUN_ON(&task_queue_);
|
|
if (file_->is_open()) {
|
|
LogToFile(std::move(rtclog_event));
|
|
} else {
|
|
LogToMemory(std::move(rtclog_event));
|
|
}
|
|
};
|
|
|
|
task_queue_.PostTask(rtc::MakeUnique<ResourceOwningTask<rtclog::Event>>(
|
|
std::move(event), event_handler));
|
|
}
|
|
|
|
bool RtcEventLogImpl::AppendEventToString(rtclog::Event* event,
|
|
ProtoString* output_string) {
|
|
RTC_DCHECK_RUN_ON(&task_queue_);
|
|
|
|
// Even though we're only serializing a single event during this call, what
|
|
// we intend to get is a list of events, with a tag and length preceding
|
|
// each actual event. To produce that, we serialize a list of a single event.
|
|
// If we later serialize additional events, the resulting ProtoString will
|
|
// be a proper concatenation of all those events.
|
|
|
|
rtclog::EventStream event_stream;
|
|
event_stream.add_stream();
|
|
|
|
// As a tweak, we swap the new event into the event-stream, write that to
|
|
// file, then swap back. This saves on some copying.
|
|
rtclog::Event* output_event = event_stream.mutable_stream(0);
|
|
output_event->Swap(event);
|
|
|
|
bool appended;
|
|
size_t potential_new_size =
|
|
written_bytes_ + output_string->size() + event_stream.ByteSize();
|
|
if (potential_new_size <= max_size_bytes_) {
|
|
event_stream.AppendToString(output_string);
|
|
appended = true;
|
|
} else {
|
|
appended = false;
|
|
}
|
|
|
|
// When the function returns, the original Event will be unchanged.
|
|
output_event->Swap(event);
|
|
|
|
return appended;
|
|
}
|
|
|
|
void RtcEventLogImpl::LogToMemory(std::unique_ptr<rtclog::Event> event) {
|
|
RTC_DCHECK_RUN_ON(&task_queue_);
|
|
RTC_DCHECK(!file_->is_open());
|
|
|
|
if (IsConfigEvent(*event.get())) {
|
|
config_history_.push_back(std::move(event));
|
|
} else {
|
|
history_.push_back(std::move(event));
|
|
if (history_.size() > kEventsInHistory) {
|
|
history_.pop_front();
|
|
}
|
|
}
|
|
}
|
|
|
|
void RtcEventLogImpl::StartLogFile() {
|
|
RTC_DCHECK_RUN_ON(&task_queue_);
|
|
RTC_DCHECK(file_->is_open());
|
|
|
|
ProtoString output_string;
|
|
|
|
// Create and serialize the LOG_START event.
|
|
// The timestamp used will correspond to when logging has started. The log
|
|
// may contain events earlier than the LOG_START event. (In general, the
|
|
// timestamps in the log are not monotonic.)
|
|
rtclog::Event start_event;
|
|
start_event.set_timestamp_us(rtc::TimeMicros());
|
|
start_event.set_type(rtclog::Event::LOG_START);
|
|
bool appended = AppendEventToString(&start_event, &output_string);
|
|
|
|
// Serialize the config information for all old streams, including streams
|
|
// which were already logged to previous files.
|
|
for (auto& event : config_history_) {
|
|
if (!appended) {
|
|
break;
|
|
}
|
|
appended = AppendEventToString(event.get(), &output_string);
|
|
}
|
|
|
|
// Serialize the events in the event queue.
|
|
while (appended && !history_.empty()) {
|
|
appended = AppendEventToString(history_.front().get(), &output_string);
|
|
if (appended) {
|
|
// Known issue - if writing to the file fails, these events will have
|
|
// been lost. If we try to open a new file, these events will be missing
|
|
// from it.
|
|
history_.pop_front();
|
|
}
|
|
}
|
|
|
|
// Write to file.
|
|
if (!file_->Write(output_string.data(), output_string.size())) {
|
|
LOG(LS_ERROR) << "FileWrapper failed to write WebRtcEventLog file.";
|
|
// The current FileWrapper implementation closes the file on error.
|
|
RTC_DCHECK(!file_->is_open());
|
|
return;
|
|
}
|
|
written_bytes_ += output_string.size();
|
|
|
|
if (!appended) {
|
|
RTC_DCHECK(file_->is_open());
|
|
StopLogFile(rtc::TimeMicros());
|
|
}
|
|
}
|
|
|
|
void RtcEventLogImpl::LogToFile(std::unique_ptr<rtclog::Event> event) {
|
|
RTC_DCHECK_RUN_ON(&task_queue_);
|
|
RTC_DCHECK(file_->is_open());
|
|
|
|
ProtoString output_string;
|
|
|
|
bool appended = AppendEventToString(event.get(), &output_string);
|
|
|
|
if (IsConfigEvent(*event.get())) {
|
|
config_history_.push_back(std::move(event));
|
|
}
|
|
|
|
if (!appended) {
|
|
RTC_DCHECK(file_->is_open());
|
|
history_.push_back(std::move(event));
|
|
StopLogFile(rtc::TimeMicros());
|
|
return;
|
|
}
|
|
|
|
// Write string to file.
|
|
if (file_->Write(output_string.data(), output_string.size())) {
|
|
written_bytes_ += output_string.size();
|
|
} else {
|
|
LOG(LS_ERROR) << "FileWrapper failed to write WebRtcEventLog file.";
|
|
// The current FileWrapper implementation closes the file on error.
|
|
RTC_DCHECK(!file_->is_open());
|
|
}
|
|
}
|
|
|
|
void RtcEventLogImpl::StopLogFile(int64_t stop_time) {
|
|
RTC_DCHECK_RUN_ON(&task_queue_);
|
|
RTC_DCHECK(file_->is_open());
|
|
|
|
ProtoString output_string;
|
|
|
|
rtclog::Event end_event;
|
|
end_event.set_timestamp_us(stop_time);
|
|
end_event.set_type(rtclog::Event::LOG_END);
|
|
bool appended = AppendEventToString(&end_event, &output_string);
|
|
|
|
if (appended) {
|
|
if (!file_->Write(output_string.data(), output_string.size())) {
|
|
LOG(LS_ERROR) << "FileWrapper failed to write WebRtcEventLog file.";
|
|
// The current FileWrapper implementation closes the file on error.
|
|
RTC_DCHECK(!file_->is_open());
|
|
}
|
|
written_bytes_ += output_string.size();
|
|
}
|
|
|
|
max_size_bytes_ = std::numeric_limits<decltype(max_size_bytes_)>::max();
|
|
written_bytes_ = 0;
|
|
|
|
file_->CloseFile();
|
|
RTC_DCHECK(!file_->is_open());
|
|
}
|
|
|
|
bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
|
|
rtclog::EventStream* result) {
|
|
char tmp_buffer[1024];
|
|
int bytes_read = 0;
|
|
std::unique_ptr<FileWrapper> dump_file(FileWrapper::Create());
|
|
if (!dump_file->OpenFile(file_name.c_str(), true)) {
|
|
return false;
|
|
}
|
|
ProtoString dump_buffer;
|
|
while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) {
|
|
dump_buffer.append(tmp_buffer, bytes_read);
|
|
}
|
|
dump_file->CloseFile();
|
|
return result->ParseFromString(dump_buffer);
|
|
}
|
|
|
|
#endif // ENABLE_RTC_EVENT_LOG
|
|
|
|
// RtcEventLog member functions.
|
|
std::unique_ptr<RtcEventLog> RtcEventLog::Create() {
|
|
#ifdef ENABLE_RTC_EVENT_LOG
|
|
// TODO(eladalon): Known issue - there's a race over |log_count_| here.
|
|
constexpr int kMaxLogCount = 5;
|
|
int count = 1 + std::atomic_fetch_add(&RtcEventLogImpl::log_count_, 1);
|
|
if (count > kMaxLogCount) {
|
|
LOG(LS_WARNING) << "Denied creation of additional WebRTC event logs. "
|
|
<< count - 1 << " logs open already.";
|
|
std::atomic_fetch_sub(&RtcEventLogImpl::log_count_, 1);
|
|
return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
|
|
}
|
|
return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl());
|
|
#else
|
|
return CreateNull();
|
|
#endif // ENABLE_RTC_EVENT_LOG
|
|
}
|
|
|
|
std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
|
|
return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
|
|
}
|
|
|
|
} // namespace webrtc
|