From 5fd6e5ec1feed9e937efbe27ba9924cee3be2b81 Mon Sep 17 00:00:00 2001 From: Elad Alon Date: Thu, 5 Oct 2017 10:19:29 +0200 Subject: [PATCH] Remove deprecated functions from RtcEventLog MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The unified Log() interface replaces the many old LogX() functions. This helps hide dependencies between the modules which log different events. TBR=stefan@webrtc.org Bug: webrtc:8111 Change-Id: I5ea9fd50ba6da87d5867513c81c5e3bdb0524a32 Reviewed-on: https://webrtc-review.googlesource.com/2689 Commit-Queue: Elad Alon Reviewed-by: Björn Terelius Reviewed-by: Elad Alon Reviewed-by: Danil Chapovalov Cr-Commit-Position: refs/heads/master@{#20159} --- .../rtc_event_log/mock/mock_rtc_event_log.h | 69 ------ logging/rtc_event_log/rtc_event_log.cc | 214 ------------------ logging/rtc_event_log/rtc_event_log.h | 171 +------------- logging/rtc_event_log/rtc_event_log_parser.h | 4 + voice_engine/channel.cc | 150 +----------- 5 files changed, 10 insertions(+), 598 deletions(-) diff --git a/logging/rtc_event_log/mock/mock_rtc_event_log.h b/logging/rtc_event_log/mock/mock_rtc_event_log.h index 1f44c0b858..6a4cdbda89 100644 --- a/logging/rtc_event_log/mock/mock_rtc_event_log.h +++ b/logging/rtc_event_log/mock/mock_rtc_event_log.h @@ -12,13 +12,8 @@ #define LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_ #include -#include #include "logging/rtc_event_log/rtc_event_log.h" -#include "logging/rtc_event_log/rtc_stream_config.h" -#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" -#include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "test/gmock.h" namespace webrtc { @@ -30,76 +25,12 @@ class MockRtcEventLog : public RtcEventLog { } MOCK_METHOD1(StartLoggingProxy, bool(RtcEventLogOutput*)); - MOCK_METHOD2(StartLogging, - bool(const std::string& file_name, int64_t max_size_bytes)); - - MOCK_METHOD2(StartLogging, - bool(rtc::PlatformFile log_file, int64_t max_size_bytes)); - MOCK_METHOD0(StopLogging, void()); virtual void Log(std::unique_ptr event) { return LogProxy(event.get()); } MOCK_METHOD1(LogProxy, void(RtcEvent*)); - - MOCK_METHOD1(LogVideoReceiveStreamConfig, - void(const rtclog::StreamConfig& config)); - - MOCK_METHOD1(LogVideoSendStreamConfig, - void(const rtclog::StreamConfig& config)); - - MOCK_METHOD1(LogAudioReceiveStreamConfig, - void(const rtclog::StreamConfig& config)); - - MOCK_METHOD1(LogAudioSendStreamConfig, - void(const rtclog::StreamConfig& config)); - MOCK_METHOD3(LogRtpHeader, - void(PacketDirection direction, - const uint8_t* header, - size_t packet_length)); - - MOCK_METHOD4(LogRtpHeader, - void(PacketDirection direction, - const uint8_t* header, - size_t packet_length, - int probe_cluster_id)); - - MOCK_METHOD3(LogRtcpPacket, - void(PacketDirection direction, - const uint8_t* packet, - size_t length)); - - MOCK_METHOD1(LogIncomingRtpHeader, void(const RtpPacketReceived& packet)); - - MOCK_METHOD2(LogOutgoingRtpHeader, - void(const RtpPacketToSend& packet, int probe_cluster_id)); - - MOCK_METHOD1(LogIncomingRtcpPacket, - void(rtc::ArrayView packet)); - - MOCK_METHOD1(LogOutgoingRtcpPacket, - void(rtc::ArrayView packet)); - - MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc)); - - MOCK_METHOD3(LogLossBasedBweUpdate, - void(int32_t bitrate_bps, - uint8_t fraction_loss, - int32_t total_packets)); - - MOCK_METHOD2(LogDelayBasedBweUpdate, - void(int32_t bitrate_bps, BandwidthUsage detector_state)); - - MOCK_METHOD1(LogAudioNetworkAdaptation, - void(const AudioEncoderRuntimeConfig& config)); - - MOCK_METHOD4(LogProbeClusterCreated, - void(int id, int bitrate_bps, int min_probes, int min_bytes)); - - MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps)); - MOCK_METHOD2(LogProbeResultFailure, - void(int id, ProbeFailureReason failure_reason)); }; } // namespace webrtc diff --git a/logging/rtc_event_log/rtc_event_log.cc b/logging/rtc_event_log/rtc_event_log.cc index 83ba029fb8..22dd2871c2 100644 --- a/logging/rtc_event_log/rtc_event_log.cc +++ b/logging/rtc_event_log/rtc_event_log.cc @@ -19,44 +19,9 @@ #include #include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h" -// TODO(eladalon): Remove events/* when the deprecated functions are removed. -#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" -#include "logging/rtc_event_log/events/rtc_event_audio_playout.h" -#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h" -#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h" -#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h" -#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h" #include "logging/rtc_event_log/events/rtc_event_logging_started.h" #include "logging/rtc_event_log/events/rtc_event_logging_stopped.h" -#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h" -#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h" -#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h" -#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" -#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h" -#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" -#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" -#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h" -#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h" -#include "logging/rtc_event_log/output/rtc_event_log_output.h" #include "logging/rtc_event_log/output/rtc_event_log_output_file.h" -#include "logging/rtc_event_log/rtc_stream_config.h" -// TODO(eladalon): Remove these when deprecated functions are removed. -#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" -#include "modules/remote_bitrate_estimator/include/bwe_defines.h" -#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "modules/rtp_rtcp/source/byte_io.h" -#include "modules/rtp_rtcp/source/rtcp_packet/app.h" -#include "modules/rtp_rtcp/source/rtcp_packet/bye.h" -#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h" -#include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" -#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" -#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h" -#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" -#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" -#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h" -#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h" -#include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "rtc_base/checks.h" #include "rtc_base/constructormagic.h" #include "rtc_base/event.h" @@ -66,7 +31,6 @@ #include "rtc_base/sequenced_task_checker.h" #include "rtc_base/task_queue.h" #include "rtc_base/thread_annotations.h" -#include "typedefs.h" // NOLINT(build/include) namespace webrtc { @@ -117,11 +81,6 @@ class RtcEventLogImpl final : public RtcEventLog { explicit RtcEventLogImpl(std::unique_ptr event_encoder); ~RtcEventLogImpl() override; - bool StartLogging(const std::string& file_name, - int64_t max_size_bytes) override; - bool StartLogging(rtc::PlatformFile platform_file, - int64_t max_size_bytes) override; - // TODO(eladalon): We should change these name to reflect that what we're // actually starting/stopping is the output of the log, not the log itself. bool StartLogging(std::unique_ptr output) override; @@ -129,47 +88,6 @@ class RtcEventLogImpl final : public RtcEventLog { void Log(std::unique_ptr event) override; - void LogVideoReceiveStreamConfig(const rtclog::StreamConfig& config) override; - void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override; - void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override; - void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override; - // TODO(terelius): This can be removed as soon as the interface has been - // updated. - void LogRtpHeader(PacketDirection direction, - const uint8_t* header, - size_t packet_length) override; - // TODO(terelius): This can be made private, non-virtual as soon as the - // interface has been updated. - void LogRtpHeader(PacketDirection direction, - const uint8_t* header, - size_t packet_length, - int probe_cluster_id) override; - void LogIncomingRtpHeader(const RtpPacketReceived& packet) override; - void LogOutgoingRtpHeader(const RtpPacketToSend& packet, - int probe_cluster_id) override; - // TODO(terelius): This can be made private, non-virtual as soon as the - // interface has been updated. - void LogRtcpPacket(PacketDirection direction, - const uint8_t* packet, - size_t length) override; - void LogIncomingRtcpPacket(rtc::ArrayView packet) override; - void LogOutgoingRtcpPacket(rtc::ArrayView packet) override; - void LogAudioPlayout(uint32_t ssrc) override; - void LogLossBasedBweUpdate(int32_t bitrate_bps, - uint8_t fraction_loss, - int32_t total_packets) override; - void LogDelayBasedBweUpdate(int32_t bitrate_bps, - BandwidthUsage detector_state) override; - void LogAudioNetworkAdaptation( - const AudioEncoderRuntimeConfig& config) override; - void LogProbeClusterCreated(int id, - int bitrate_bps, - int min_probes, - int min_bytes) override; - void LogProbeResultSuccess(int id, int bitrate_bps) override; - void LogProbeResultFailure(int id, - ProbeFailureReason failure_reason) override; - private: // Appends an event to the output protobuf string, returning true on success. // Fails and returns false in case the limit on output size prevents the @@ -232,20 +150,6 @@ RtcEventLogImpl::~RtcEventLogImpl() { RTC_DCHECK_GE(count, 0); } -bool RtcEventLogImpl::StartLogging(const std::string& file_name, - int64_t max_size_bytes) { - RTC_CHECK(max_size_bytes > 0 || max_size_bytes == kUnlimitedOutput); - return StartLogging(rtc::MakeUnique( - file_name, rtc::saturated_cast(max_size_bytes))); -} - -bool RtcEventLogImpl::StartLogging(rtc::PlatformFile platform_file, - int64_t max_size_bytes) { - RTC_CHECK(max_size_bytes > 0 || max_size_bytes == kUnlimitedOutput); - return StartLogging(rtc::MakeUnique( - platform_file, rtc::saturated_cast(max_size_bytes))); -} - bool RtcEventLogImpl::StartLogging(std::unique_ptr output) { RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_); @@ -310,124 +214,6 @@ void RtcEventLogImpl::Log(std::unique_ptr event) { std::move(event), event_handler)); } -void RtcEventLogImpl::LogVideoReceiveStreamConfig( - const rtclog::StreamConfig& config) { - Log(rtc::MakeUnique( - rtc::MakeUnique(config))); -} - -void RtcEventLogImpl::LogVideoSendStreamConfig( - const rtclog::StreamConfig& config) { - Log(rtc::MakeUnique( - rtc::MakeUnique(config))); -} - -void RtcEventLogImpl::LogAudioReceiveStreamConfig( - const rtclog::StreamConfig& config) { - Log(rtc::MakeUnique( - rtc::MakeUnique(config))); -} - -void RtcEventLogImpl::LogAudioSendStreamConfig( - const rtclog::StreamConfig& config) { - Log(rtc::MakeUnique( - rtc::MakeUnique(config))); -} - -void RtcEventLogImpl::LogIncomingRtpHeader(const RtpPacketReceived& packet) { - Log(rtc::MakeUnique(packet)); -} - -void RtcEventLogImpl::LogOutgoingRtpHeader(const RtpPacketToSend& packet, - int probe_cluster_id) { - Log(rtc::MakeUnique(packet, probe_cluster_id)); -} - -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) { - // TODO(eladalon): This is highly inefficient. We're only doing this for - // the deprecated interface. We should remove this soon. - if (direction == PacketDirection::kIncomingPacket) { - RtpPacketReceived packet; - packet.Parse(header, packet_length); - Log(rtc::MakeUnique(packet)); - } else { - RTC_CHECK_EQ(direction, PacketDirection::kOutgoingPacket); - RtpPacketToSend packet(nullptr); - packet.Parse(header, packet_length); - Log(rtc::MakeUnique(packet, probe_cluster_id)); - } -} - -void RtcEventLogImpl::LogIncomingRtcpPacket( - rtc::ArrayView packet) { - Log(rtc::MakeUnique(packet)); -} - -void RtcEventLogImpl::LogOutgoingRtcpPacket( - rtc::ArrayView packet) { - Log(rtc::MakeUnique(packet)); -} - -void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction, - const uint8_t* packet, - size_t length) { - if (direction == PacketDirection::kIncomingPacket) { - LogIncomingRtcpPacket(rtc::ArrayView(packet, length)); - } else { - RTC_CHECK_EQ(direction, PacketDirection::kOutgoingPacket); - LogOutgoingRtcpPacket(rtc::ArrayView(packet, length)); - } -} - -void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) { - Log(rtc::MakeUnique(ssrc)); -} - -void RtcEventLogImpl::LogLossBasedBweUpdate(int32_t bitrate_bps, - uint8_t fraction_loss, - int32_t total_packets) { - Log(rtc::MakeUnique(bitrate_bps, fraction_loss, - total_packets)); -} - -void RtcEventLogImpl::LogDelayBasedBweUpdate(int32_t bitrate_bps, - BandwidthUsage detector_state) { - Log(rtc::MakeUnique(bitrate_bps, - detector_state)); -} - -void RtcEventLogImpl::LogAudioNetworkAdaptation( - const AudioEncoderRuntimeConfig& config) { - Log(rtc::MakeUnique( - rtc::MakeUnique(config))); -} - -void RtcEventLogImpl::LogProbeClusterCreated(int id, - int bitrate_bps, - int min_probes, - int min_bytes) { - Log(rtc::MakeUnique(id, bitrate_bps, min_probes, - min_bytes)); -} - -void RtcEventLogImpl::LogProbeResultSuccess(int id, int bitrate_bps) { - Log(rtc::MakeUnique(id, bitrate_bps)); -} - -void RtcEventLogImpl::LogProbeResultFailure(int id, - ProbeFailureReason failure_reason) { - Log(rtc::MakeUnique(id, failure_reason)); -} - bool RtcEventLogImpl::AppendEventToString(const RtcEvent& event, std::string* output_string) { RTC_DCHECK_RUN_ON(&task_queue_); diff --git a/logging/rtc_event_log/rtc_event_log.h b/logging/rtc_event_log/rtc_event_log.h index cb9d097888..f798045729 100644 --- a/logging/rtc_event_log/rtc_event_log.h +++ b/logging/rtc_event_log/rtc_event_log.h @@ -14,33 +14,13 @@ #include #include -// TODO(eladalon): Remove this include once LogIncomingRtcpPacket(), etc., have -// been removed (they are currently deprecated). -#include "api/array_view.h" -#include "common_types.h" // NOLINT(build/include) +#include "logging/rtc_event_log/events/rtc_event.h" #include "logging/rtc_event_log/output/rtc_event_log_output.h" -// TODO(eladalon): This is here because of ProbeFailureReason; remove this -// dependency along with the deprecated LogProbeResultFailure(). -#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h" -// TODO(eladalon): Remove this #include once the deprecated versions of -// StartLogging() have been removed. -#include "rtc_base/platform_file.h" namespace webrtc { -namespace rtclog { -class EventStream; // Storage class automatically generated from protobuf. -// TODO(eladalon): Get rid of this when deprecated methods are removed. -struct StreamConfig; -} // namespace rtclog - class Clock; -// TODO(eladalon): The following may be removed when the deprecated methods -// are removed. -struct AudioEncoderRuntimeConfig; -class RtpPacketReceived; -class RtpPacketToSend; -enum class BandwidthUsage; + enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket }; class RtcEventLog { @@ -55,16 +35,12 @@ class RtcEventLog { virtual ~RtcEventLog() {} // Factory method to create an RtcEventLog object. - // TODO(eladalon): Get rid of the default value after internal projects fixed. - static std::unique_ptr Create( - EncodingType encoding_type = EncodingType::Legacy); + static std::unique_ptr Create(EncodingType encoding_type); // TODO(nisse): webrtc::Clock is deprecated. Delete this method and // above forward declaration of Clock when // webrtc/system_wrappers/include/clock.h is deleted. - // TODO(eladalon): Get rid of the default value after internal projects fixed. - static std::unique_ptr Create( - const Clock* clock, - EncodingType encoding_type = EncodingType::Legacy) { + static std::unique_ptr Create(const Clock* clock, + EncodingType encoding_type) { return Create(encoding_type); } @@ -75,115 +51,12 @@ class RtcEventLog { // and may close itself once it has reached the maximum size. virtual bool StartLogging(std::unique_ptr output) = 0; - // Starts logging a maximum of max_size_bytes bytes to the specified file. - // If the file already exists it will be overwritten. - // If max_size_bytes <= 0, logging will be active until StopLogging is called. - // The function has no effect and returns false if we can't start a new log - // e.g. because we are already logging or the file cannot be opened. - RTC_DEPRECATED virtual bool StartLogging(const std::string& file_name, - int64_t max_size_bytes) = 0; - - // Same as above. The RtcEventLog takes ownership of the file if the call - // is successful, i.e. if it returns true. - RTC_DEPRECATED virtual bool StartLogging(rtc::PlatformFile platform_file, - int64_t max_size_bytes) = 0; - - // Deprecated. Pass an explicit file size limit. - RTC_DEPRECATED bool StartLogging(const std::string& file_name) { - return StartLogging(file_name, 10000000); - } - - // Deprecated. Pass an explicit file size limit. - RTC_DEPRECATED bool StartLogging(rtc::PlatformFile platform_file) { - return StartLogging(platform_file, 10000000); - } - // Stops logging to file and waits until the file has been closed, after // which it would be permissible to read and/or modify it. virtual void StopLogging() = 0; // Log an RTC event (the type of event is determined by the subclass). virtual void Log(std::unique_ptr event) = 0; - - // Logs configuration information for a video receive stream. - RTC_DEPRECATED virtual void LogVideoReceiveStreamConfig( - const rtclog::StreamConfig& config) = 0; - - // Logs configuration information for a video send stream. - RTC_DEPRECATED virtual void LogVideoSendStreamConfig( - const rtclog::StreamConfig& config) = 0; - - // Logs configuration information for an audio receive stream. - RTC_DEPRECATED virtual void LogAudioReceiveStreamConfig( - const rtclog::StreamConfig& config) = 0; - - // Logs configuration information for an audio send stream. - RTC_DEPRECATED virtual void LogAudioSendStreamConfig( - const rtclog::StreamConfig& config) = 0; - - RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction, - const uint8_t* header, - size_t packet_length) {} - - RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction, - const uint8_t* header, - size_t packet_length, - int probe_cluster_id) {} - - // Logs the header of an incoming RTP packet. |packet_length| - // is the total length of the packet, including both header and payload. - RTC_DEPRECATED virtual void LogIncomingRtpHeader( - const RtpPacketReceived& packet) = 0; - - // Logs the header of an incoming RTP packet. |packet_length| - // is the total length of the packet, including both header and payload. - RTC_DEPRECATED virtual void LogOutgoingRtpHeader( - const RtpPacketToSend& packet, - int probe_cluster_id) = 0; - - RTC_DEPRECATED virtual void LogRtcpPacket(PacketDirection direction, - const uint8_t* header, - size_t packet_length) {} - - // Logs an incoming RTCP packet. - RTC_DEPRECATED virtual void LogIncomingRtcpPacket( - rtc::ArrayView packet) = 0; - - // Logs an outgoing RTCP packet. - RTC_DEPRECATED virtual void LogOutgoingRtcpPacket( - rtc::ArrayView packet) = 0; - - // Logs an audio playout event. - RTC_DEPRECATED virtual void LogAudioPlayout(uint32_t ssrc) = 0; - - // Logs a bitrate update from the bandwidth estimator based on packet loss. - RTC_DEPRECATED virtual void LogLossBasedBweUpdate(int32_t bitrate_bps, - uint8_t fraction_loss, - int32_t total_packets) = 0; - - // Logs a bitrate update from the bandwidth estimator based on delay changes. - RTC_DEPRECATED virtual void LogDelayBasedBweUpdate( - int32_t bitrate_bps, - BandwidthUsage detector_state) = 0; - - // Logs audio encoder re-configuration driven by audio network adaptor. - RTC_DEPRECATED virtual void LogAudioNetworkAdaptation( - const AudioEncoderRuntimeConfig& config) = 0; - - // Logs when a probe cluster is created. - RTC_DEPRECATED virtual void LogProbeClusterCreated(int id, - int bitrate_bps, - int min_probes, - int min_bytes) = 0; - - // Logs the result of a successful probing attempt. - RTC_DEPRECATED virtual void LogProbeResultSuccess(int id, - int bitrate_bps) = 0; - - // Logs the result of an unsuccessful probing attempt. - RTC_DEPRECATED virtual void LogProbeResultFailure( - int id, - ProbeFailureReason failure_reason) = 0; }; // No-op implementation is used if flag is not set, or in tests. @@ -192,42 +65,8 @@ class RtcEventLogNullImpl : public RtcEventLog { bool StartLogging(std::unique_ptr output) override { return false; } - bool StartLogging(const std::string& file_name, - int64_t max_size_bytes) override { - return false; - } - bool StartLogging(rtc::PlatformFile platform_file, - int64_t max_size_bytes) override { - return false; - } void StopLogging() override {} void Log(std::unique_ptr event) override {} - void LogVideoReceiveStreamConfig( - const rtclog::StreamConfig& config) override {} - void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {} - void LogAudioReceiveStreamConfig( - const rtclog::StreamConfig& config) override {} - void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {} - void LogIncomingRtpHeader(const RtpPacketReceived& packet) override {} - void LogOutgoingRtpHeader(const RtpPacketToSend& packet, - int probe_cluster_id) override {} - void LogIncomingRtcpPacket(rtc::ArrayView packet) override {} - void LogOutgoingRtcpPacket(rtc::ArrayView packet) override {} - void LogAudioPlayout(uint32_t ssrc) override {} - void LogLossBasedBweUpdate(int32_t bitrate_bps, - uint8_t fraction_loss, - int32_t total_packets) override {} - void LogDelayBasedBweUpdate(int32_t bitrate_bps, - BandwidthUsage detector_state) override {} - void LogAudioNetworkAdaptation( - const AudioEncoderRuntimeConfig& config) override {} - void LogProbeClusterCreated(int id, - int bitrate_bps, - int min_probes, - int min_bytes) override{}; - void LogProbeResultSuccess(int id, int bitrate_bps) override{}; - void LogProbeResultFailure(int id, - ProbeFailureReason failure_reason) override{}; }; } // namespace webrtc diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h index 49809594c1..8192c30b98 100644 --- a/logging/rtc_event_log/rtc_event_log_parser.h +++ b/logging/rtc_event_log/rtc_event_log_parser.h @@ -17,6 +17,7 @@ #include "call/video_receive_stream.h" #include "call/video_send_stream.h" +#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h" #include "logging/rtc_event_log/rtc_event_log.h" #include "logging/rtc_event_log/rtc_stream_config.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" @@ -34,8 +35,11 @@ RTC_POP_IGNORING_WUNDEF() namespace webrtc { +enum class BandwidthUsage; enum class MediaType; +struct AudioEncoderRuntimeConfig; + class ParsedRtcEventLog { friend class RtcEventLogTestHelper; diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc index 19873f1c17..542ea2d831 100644 --- a/voice_engine/channel.cc +++ b/voice_engine/channel.cc @@ -12,6 +12,7 @@ #include #include +#include #include #include #include @@ -20,24 +21,7 @@ #include "audio/utility/audio_frame_operations.h" #include "call/rtp_transport_controller_send_interface.h" #include "logging/rtc_event_log/rtc_event_log.h" -// TODO(eladalon): Remove events/* after removing the deprecated functions. -#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" #include "logging/rtc_event_log/events/rtc_event_audio_playout.h" -#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h" -#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h" -#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h" -#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h" -#include "logging/rtc_event_log/events/rtc_event_logging_started.h" -#include "logging/rtc_event_log/events/rtc_event_logging_stopped.h" -#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h" -#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h" -#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h" -#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" -#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h" -#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" -#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" -#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h" -#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "modules/audio_coding/codecs/audio_format_conversion.h" #include "modules/audio_device/include/audio_device.h" @@ -86,18 +70,6 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { return false; } - bool StartLogging(const std::string& file_name, - int64_t max_size_bytes) override { - RTC_NOTREACHED(); - return false; - } - - bool StartLogging(rtc::PlatformFile log_file, - int64_t max_size_bytes) override { - RTC_NOTREACHED(); - return false; - } - void StopLogging() override { RTC_NOTREACHED(); } void Log(std::unique_ptr event) override { @@ -107,126 +79,6 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { } } - void LogVideoReceiveStreamConfig( - const webrtc::rtclog::StreamConfig&) override { - RTC_NOTREACHED(); - } - - void LogVideoSendStreamConfig(const webrtc::rtclog::StreamConfig&) override { - RTC_NOTREACHED(); - } - - void LogAudioReceiveStreamConfig( - const webrtc::rtclog::StreamConfig& config) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique( - rtc::MakeUnique(config))); - } - } - - void LogAudioSendStreamConfig( - const webrtc::rtclog::StreamConfig& config) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique( - rtc::MakeUnique(config))); - } - } - - void LogIncomingRtpHeader(const RtpPacketReceived& packet) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique(packet)); - } - } - - void LogOutgoingRtpHeader(const RtpPacketToSend& packet, - int probe_cluster_id) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log( - rtc::MakeUnique(packet, probe_cluster_id)); - } - } - - void LogIncomingRtcpPacket(rtc::ArrayView packet) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique(packet)); - } - } - - void LogOutgoingRtcpPacket(rtc::ArrayView packet) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique(packet)); - } - } - - void LogAudioPlayout(uint32_t ssrc) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique(ssrc)); - } - } - - void LogLossBasedBweUpdate(int32_t bitrate_bps, - uint8_t fraction_loss, - int32_t total_packets) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique( - bitrate_bps, fraction_loss, total_packets)); - } - } - - void LogDelayBasedBweUpdate(int32_t bitrate_bps, - BandwidthUsage detector_state) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique( - bitrate_bps, detector_state)); - } - } - - void LogAudioNetworkAdaptation( - const AudioEncoderRuntimeConfig& config) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique( - rtc::MakeUnique(config))); - } - } - - void LogProbeClusterCreated(int id, - int bitrate_bps, - int min_probes, - int min_bytes) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log(rtc::MakeUnique( - id, bitrate_bps, min_probes, min_bytes)); - } - }; - - void LogProbeResultSuccess(int id, int bitrate_bps) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log( - rtc::MakeUnique(id, bitrate_bps)); - } - }; - - void LogProbeResultFailure(int id, - ProbeFailureReason failure_reason) override { - rtc::CritScope lock(&crit_); - if (event_log_) { - event_log_->Log( - rtc::MakeUnique(id, failure_reason)); - } - }; - void SetEventLog(RtcEventLog* event_log) { rtc::CritScope lock(&crit_); event_log_ = event_log;