Reland "Remove deprecated functions from RtcEventLog"
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: I36c8b6c4cf03d738c9033af2e98db6dc200eede9
Reviewed-on: https://webrtc-review.googlesource.com/6940
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20170}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 1f44c0b..6a4cdbd 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 <memory>
-#include <string>
#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 @@
}
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<RtcEvent> 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<const uint8_t> packet));
-
- MOCK_METHOD1(LogOutgoingRtcpPacket,
- void(rtc::ArrayView<const uint8_t> 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 83ba029..22dd287 100644
--- a/logging/rtc_event_log/rtc_event_log.cc
+++ b/logging/rtc_event_log/rtc_event_log.cc
@@ -19,44 +19,9 @@
#include <vector>
#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 @@
explicit RtcEventLogImpl(std::unique_ptr<RtcEventLogEncoder> 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<RtcEventLogOutput> output) override;
@@ -129,47 +88,6 @@
void Log(std::unique_ptr<RtcEvent> 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<const uint8_t> packet) override;
- void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> 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 @@
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<RtcEventLogOutputFile>(
- file_name, rtc::saturated_cast<size_t>(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<RtcEventLogOutputFile>(
- platform_file, rtc::saturated_cast<size_t>(max_size_bytes)));
-}
-
bool RtcEventLogImpl::StartLogging(std::unique_ptr<RtcEventLogOutput> output) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
@@ -310,124 +214,6 @@
std::move(event), event_handler));
}
-void RtcEventLogImpl::LogVideoReceiveStreamConfig(
- const rtclog::StreamConfig& config) {
- Log(rtc::MakeUnique<RtcEventVideoReceiveStreamConfig>(
- rtc::MakeUnique<rtclog::StreamConfig>(config)));
-}
-
-void RtcEventLogImpl::LogVideoSendStreamConfig(
- const rtclog::StreamConfig& config) {
- Log(rtc::MakeUnique<RtcEventVideoSendStreamConfig>(
- rtc::MakeUnique<rtclog::StreamConfig>(config)));
-}
-
-void RtcEventLogImpl::LogAudioReceiveStreamConfig(
- const rtclog::StreamConfig& config) {
- Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>(
- rtc::MakeUnique<rtclog::StreamConfig>(config)));
-}
-
-void RtcEventLogImpl::LogAudioSendStreamConfig(
- const rtclog::StreamConfig& config) {
- Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>(
- rtc::MakeUnique<rtclog::StreamConfig>(config)));
-}
-
-void RtcEventLogImpl::LogIncomingRtpHeader(const RtpPacketReceived& packet) {
- Log(rtc::MakeUnique<RtcEventRtpPacketIncoming>(packet));
-}
-
-void RtcEventLogImpl::LogOutgoingRtpHeader(const RtpPacketToSend& packet,
- int probe_cluster_id) {
- Log(rtc::MakeUnique<RtcEventRtpPacketOutgoing>(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<RtcEventRtpPacketIncoming>(packet));
- } else {
- RTC_CHECK_EQ(direction, PacketDirection::kOutgoingPacket);
- RtpPacketToSend packet(nullptr);
- packet.Parse(header, packet_length);
- Log(rtc::MakeUnique<RtcEventRtpPacketOutgoing>(packet, probe_cluster_id));
- }
-}
-
-void RtcEventLogImpl::LogIncomingRtcpPacket(
- rtc::ArrayView<const uint8_t> packet) {
- Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>(packet));
-}
-
-void RtcEventLogImpl::LogOutgoingRtcpPacket(
- rtc::ArrayView<const uint8_t> packet) {
- Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet));
-}
-
-void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
- const uint8_t* packet,
- size_t length) {
- if (direction == PacketDirection::kIncomingPacket) {
- LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t>(packet, length));
- } else {
- RTC_CHECK_EQ(direction, PacketDirection::kOutgoingPacket);
- LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t>(packet, length));
- }
-}
-
-void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
- Log(rtc::MakeUnique<RtcEventAudioPlayout>(ssrc));
-}
-
-void RtcEventLogImpl::LogLossBasedBweUpdate(int32_t bitrate_bps,
- uint8_t fraction_loss,
- int32_t total_packets) {
- Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>(bitrate_bps, fraction_loss,
- total_packets));
-}
-
-void RtcEventLogImpl::LogDelayBasedBweUpdate(int32_t bitrate_bps,
- BandwidthUsage detector_state) {
- Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>(bitrate_bps,
- detector_state));
-}
-
-void RtcEventLogImpl::LogAudioNetworkAdaptation(
- const AudioEncoderRuntimeConfig& config) {
- Log(rtc::MakeUnique<RtcEventAudioNetworkAdaptation>(
- rtc::MakeUnique<AudioEncoderRuntimeConfig>(config)));
-}
-
-void RtcEventLogImpl::LogProbeClusterCreated(int id,
- int bitrate_bps,
- int min_probes,
- int min_bytes) {
- Log(rtc::MakeUnique<RtcEventProbeClusterCreated>(id, bitrate_bps, min_probes,
- min_bytes));
-}
-
-void RtcEventLogImpl::LogProbeResultSuccess(int id, int bitrate_bps) {
- Log(rtc::MakeUnique<RtcEventProbeResultSuccess>(id, bitrate_bps));
-}
-
-void RtcEventLogImpl::LogProbeResultFailure(int id,
- ProbeFailureReason failure_reason) {
- Log(rtc::MakeUnique<RtcEventProbeResultFailure>(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 cb9d097..f798045 100644
--- a/logging/rtc_event_log/rtc_event_log.h
+++ b/logging/rtc_event_log/rtc_event_log.h
@@ -14,33 +14,13 @@
#include <memory>
#include <string>
-// 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 @@
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<RtcEventLog> Create(
- EncodingType encoding_type = EncodingType::Legacy);
+ static std::unique_ptr<RtcEventLog> 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<RtcEventLog> Create(
- const Clock* clock,
- EncodingType encoding_type = EncodingType::Legacy) {
+ static std::unique_ptr<RtcEventLog> Create(const Clock* clock,
+ EncodingType encoding_type) {
return Create(encoding_type);
}
@@ -75,115 +51,12 @@
// and may close itself once it has reached the maximum size.
virtual bool StartLogging(std::unique_ptr<RtcEventLogOutput> 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<RtcEvent> 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<const uint8_t> packet) = 0;
-
- // Logs an outgoing RTCP packet.
- RTC_DEPRECATED virtual void LogOutgoingRtcpPacket(
- rtc::ArrayView<const uint8_t> 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 @@
bool StartLogging(std::unique_ptr<RtcEventLogOutput> 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<RtcEvent> 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<const uint8_t> packet) override {}
- void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> 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 4980959..8192c30 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 @@
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 19873f1..542ea2d 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -12,6 +12,7 @@
#include <algorithm>
#include <map>
+#include <memory>
#include <string>
#include <utility>
#include <vector>
@@ -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 @@
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<RtcEvent> event) override {
@@ -107,126 +79,6 @@
}
}
- 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<RtcEventAudioReceiveStreamConfig>(
- rtc::MakeUnique<webrtc::rtclog::StreamConfig>(config)));
- }
- }
-
- void LogAudioSendStreamConfig(
- const webrtc::rtclog::StreamConfig& config) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>(
- rtc::MakeUnique<webrtc::rtclog::StreamConfig>(config)));
- }
- }
-
- void LogIncomingRtpHeader(const RtpPacketReceived& packet) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(rtc::MakeUnique<RtcEventRtpPacketIncoming>(packet));
- }
- }
-
- void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
- int probe_cluster_id) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(
- rtc::MakeUnique<RtcEventRtpPacketOutgoing>(packet, probe_cluster_id));
- }
- }
-
- void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>(packet));
- }
- }
-
- void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet));
- }
- }
-
- void LogAudioPlayout(uint32_t ssrc) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(rtc::MakeUnique<RtcEventAudioPlayout>(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<RtcEventBweUpdateLossBased>(
- 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<RtcEventBweUpdateDelayBased>(
- bitrate_bps, detector_state));
- }
- }
-
- void LogAudioNetworkAdaptation(
- const AudioEncoderRuntimeConfig& config) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(rtc::MakeUnique<RtcEventAudioNetworkAdaptation>(
- rtc::MakeUnique<AudioEncoderRuntimeConfig>(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<RtcEventProbeClusterCreated>(
- 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<RtcEventProbeResultSuccess>(id, bitrate_bps));
- }
- };
-
- void LogProbeResultFailure(int id,
- ProbeFailureReason failure_reason) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->Log(
- rtc::MakeUnique<RtcEventProbeResultFailure>(id, failure_reason));
- }
- };
-
void SetEventLog(RtcEventLog* event_log) {
rtc::CritScope lock(&crit_);
event_log_ = event_log;