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Reland "New protobuf format for event log.""""

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This reverts commit 6cfbc35ad7e6c64874c1e2dbd58b8d7c4ab3a679.

Reason for revert: Fixing downstream projects.

Original change's description:
> Revert "Revert "Revert "New protobuf format for event log."""
> 
> This reverts commit ef8f42040367b3809295a007d7eeeff4526e1b39.
> 
> Reason for revert: New problems with downstream project.
> 
> Original change's description:
> > Revert "Revert "New protobuf format for event log.""
> > 
> > This reverts commit 546373fc66e24d041e8eb8ffd2fc522847d841d1.
> > 
> > Reason for revert: Downstream project fixed.
> > 
> > Original change's description:
> > > Revert "New protobuf format for event log."
> > > 
> > > This reverts commit 99463c14dbbc88732f0991cb30e7bbfcdaeb3cdc.
> > > 
> > > Reason for revert: Speculative revert for downstream project breakage.
> > > 
> > > Original change's description:
> > > > New protobuf format for event log.
> > > > 
> > > > Bug: webrtc:6295
> > > > Change-Id: Ie20a2808a4f076b05fb6195f4fed73215f6fd3b2
> > > > Reviewed-on: https://webrtc-review.googlesource.com/8880
> > > > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > > > Reviewed-by: Per Kjellander <perkj@webrtc.org>
> > > > Reviewed-by: Dino Radaković <dinor@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#21291}
> > > 
> > > TBR=terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org
> > > 
> > > Change-Id: Ic319170a7a777002ca29248d102cb4e26966d5ae
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:6295
> > > Reviewed-on: https://webrtc-review.googlesource.com/33400
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21292}
> > 
> > TBR=brandtr@webrtc.org,terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org
> > 
> > Change-Id: I9e96e5007d0447e63178d47c7330488b2a8f2b6f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:6295
> > Reviewed-on: https://webrtc-review.googlesource.com/33440
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21296}
> 
> TBR=brandtr@webrtc.org,terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org
> 
> Change-Id: I4eb15c809f67af13ffa7b7df6eb06088af21f63f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:6295
> Reviewed-on: https://webrtc-review.googlesource.com/33480
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21297}

TBR=brandtr@webrtc.org,terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org

Change-Id: I7895575f2b6e4ec2c36296fe81a7596147158601
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6295
Reviewed-on: https://webrtc-review.googlesource.com/33520
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21299}
This commit is contained in:
Mirko Bonadei
2017-12-15 17:12:42 +01:00
committed by Commit Bot
parent eb0edd832a
commit b3c210fa56
2 changed files with 401 additions and 0 deletions
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8
logging/BUILD.gn
View File

@ -137,6 +137,13 @@ if (rtc_enable_protobuf) {
proto_out_dir = "logging/rtc_event_log"
}
proto_library("rtc_event_log2_proto") {
sources = [
"rtc_event_log/rtc_event_log2.proto",
]
proto_out_dir = "logging/rtc_event_log"
}
rtc_static_library("rtc_event_log_parser") {
sources = [
"rtc_event_log/rtc_event_log_parser.cc",
@ -144,6 +151,7 @@ if (rtc_enable_protobuf) {
]
deps = [
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_proto",
"..:webrtc_common",

393
logging/rtc_event_log/rtc_event_log2.proto Normal file
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@ -0,0 +1,393 @@
// THIS FILE IS EXPERIMENTAL. BREAKING CHANGES MAY BE MADE AT ANY TIME
// WITHOUT PRIOR WARNING. THIS FILE SHOULD NOT BE USED IN PRODUCTION CODE.
syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.rtclog2;
// At the top level, a WebRTC event log is just an EventStream object. Note that
// concatenating multiple EventStreams in the same file is equivalent to a
// single EventStream object containing the same events. Hence, it is not
// necessary to wait for the entire log to be complete before beginning to
// write it to a file.
message EventStream {
// Deprecated - Maintained for compatibility with the old event log.
// TODO(terelius): Maybe we can remove this and instead check the stream for
// presence of a version field. That requires a custom protobuf parser, but we
// have that already anyway.
repeated Event stream = 1 [deprecated = true];
// required - The version number must be 2 for this version of the event log.
optional uint32 version = 2;
repeated IncomingRtpPackets incoming_rtp_packets = 3;
repeated OutgoingRtpPackets outgoing_rtp_packets = 4;
repeated IncomingRtcpPackets incoming_rtcp_packets = 5;
repeated OutgoingRtcpPackets outgoing_rtcp_packets = 6;
repeated AudioPlayoutEvents audio_playout_events = 7;
// The field tags 8-15 are reserved for the most common events
repeated BeginLogEvent begin_log_events = 16;
repeated EndLogEvent end_log_events = 17;
repeated LossBasedBweUpdates loss_based_bwe_updates = 18;
repeated DelayBasedBweUpdates delay_based_bwe_updates = 19;
repeated AudioNetworkAdaptations audio_network_adaptations = 20;
repeated BweProbeCluster probe_clusters = 21;
repeated BweProbeResultSuccess probe_success = 22;
repeated BweProbeResultFailure probe_failure = 23;
repeated AudioRecvStreamConfig audio_recv_stream_configs = 101;
repeated AudioSendStreamConfig audio_send_stream_configs = 102;
repeated VideoRecvStreamConfig video_recv_stream_configs = 103;
repeated VideoSendStreamConfig video_send_stream_configs = 104;
}
// DEPRECATED.
message Event {
// TODO(terelius): Do we want to preserve the old Event definition here?
}
message IncomingRtpPackets {
optional int64 timestamp_ms = 1;
// RTP marker bit, used to label boundaries within e.g. video frames.
optional bool marker = 2;
// RTP payload type.
optional uint32 payload_type = 3;
// RTP sequence number.
optional uint32 sequence_number = 4;
// RTP monotonic clock timestamp (not actual time).
optional fixed32 rtp_timestamp = 5;
// Synchronization source of this packet's RTP stream.
optional fixed32 ssrc = 6;
// TODO(terelius/dinor): Add CSRCs. Field number 7 reserved for this purpose.
// required - The size of the packet including both payload and header.
optional uint32 packet_size = 8;
// Optional header extensions.
optional int32 transmission_time_offset = 9;
optional uint32 absolute_send_time = 10;
optional uint32 transport_sequence_number = 11;
optional uint32 audio_level = 12;
// TODO(terelius): Add header extensions like video rotation, playout delay?
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes marker_deltas = 102;
optional bytes payload_type_deltas = 103;
optional bytes sequence_number_deltas = 104;
optional bytes rtp_timestamp_deltas = 105;
optional bytes ssrc_deltas = 106;
optional bytes packet_size_deltas = 107;
optional bytes transmission_time_offset_deltas = 108;
optional bytes absolute_send_time_deltas = 109;
optional bytes transport_sequence_number_deltas = 110;
optional bytes audio_level_deltas = 111;
}
message OutgoingRtpPackets {
optional int64 timestamp_ms = 1;
// RTP marker bit, used to label boundaries within e.g. video frames.
optional bool marker = 2;
optional uint32 payload_type = 3;
// RTP sequence number.
optional uint32 sequence_number = 4;
// RTP monotonic clock timestamp (not actual time).
optional fixed32 rtp_timestamp = 5;
// Synchronization source of this packet's RTP stream.
optional fixed32 ssrc = 6;
// TODO(terelius/dinor): Add CSRCs. Field number 7 reserved for this purpose.
// required - The size of the packet including both payload and header.
optional uint32 packet_size = 8;
// Optional header extensions.
optional int32 transmission_time_offset = 9;
optional uint32 absolute_send_time = 10;
optional uint32 transport_sequence_number = 11;
optional uint32 audio_level = 12;
// TODO(terelius): Add header extensions like video rotation, playout delay?
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes marker_deltas = 102;
optional bytes payload_type_deltas = 103;
optional bytes sequence_number_deltas = 104;
optional bytes rtp_timestamp_deltas = 105;
optional bytes ssrc_deltas = 106;
optional bytes packet_size_deltas = 107;
optional bytes probe_cluster_id_deltas = 108;
optional bytes transmission_time_offset_deltas = 109;
optional bytes absolute_send_time_deltas = 110;
optional bytes transport_sequence_number_deltas = 111;
}
message IncomingRtcpPackets {
optional int64 timestamp_ms = 1;
// required - The whole packet including both payload and header.
optional bytes raw_packet = 2;
// TODO(terelius): Feasible to log parsed RTCP instead?
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes raw_packet_deltas = 102;
}
message OutgoingRtcpPackets {
optional int64 timestamp_ms = 1;
// required - The whole packet including both payload and header.
optional bytes raw_packet = 2;
// TODO(terelius): Feasible to log parsed RTCP instead?
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes raw_packet_deltas = 102;
}
message AudioPlayoutEvents {
optional int64 timestamp_ms = 1;
// required - The SSRC of the audio stream associated with the playout event.
optional uint32 local_ssrc = 2;
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes local_ssrc_deltas = 102;
}
message BeginLogEvent {
optional int64 timestamp_ms = 1;
}
message EndLogEvent {
optional int64 timestamp_ms = 1;
}
message LossBasedBweUpdates {
optional int64 timestamp_ms = 1;
// TODO(terelius): Update log interface to unsigned.
// required - Bandwidth estimate (in bps) after the update.
optional uint32 bitrate_bps = 2;
// required - Fraction of lost packets since last receiver report
// computed as floor( 256 * (#lost_packets / #total_packets) ).
// The possible values range from 0 to 255.
optional uint32 fraction_loss = 3;
// TODO(terelius): Is this really needed? Remove or make optional?
// TODO(terelius): Update log interface to unsigned.
// required - Total number of packets that the BWE update is based on.
optional uint32 total_packets = 4;
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes bitrate_deltas_bps = 102;
optional bytes fraction_loss_deltas = 103;
optional bytes total_packets_deltas = 104;
}
message DelayBasedBweUpdates {
optional int64 timestamp_ms = 1;
// required - Bandwidth estimate (in bps) after the update.
optional uint32 bitrate_bps = 2;
enum DetectorState {
BWE_NORMAL = 0;
BWE_UNDERUSING = 1;
BWE_OVERUSING = 2;
}
optional DetectorState detector_state = 3;
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes bitrate_deltas_bps = 102;
optional bytes detector_state_deltas = 103;
}
// Maps RTP header extension names to numerical IDs.
message RtpHeaderExtensionConfig {
// Optional IDs for the header extensions. Each ID is a 4-bit number that is
// only set if that extension is configured.
// TODO(terelius): Can we skip transmission_time_offset? When is it used?
optional int32 transmission_time_offset_id = 1;
optional int32 absolute_send_time_id = 2;
optional int32 transport_sequence_number_id = 3;
optional int32 audio_level_id = 4;
// TODO(terelius): Add video_rotation and playout delay?
}
message VideoRecvStreamConfig {
optional int64 timestamp_ms = 1;
// required - Synchronization source (stream identifier) to be received.
optional uint32 remote_ssrc = 2;
// required - Sender SSRC used for sending RTCP (such as receiver reports).
optional uint32 local_ssrc = 3;
// required if RTX is configured
optional uint32 rtx_ssrc = 4;
// optional - RTP source stream ID
optional bytes rsid = 5;
// IDs for the header extension we care about. Only required if there are
// header extensions configured.
optional RtpHeaderExtensionConfig header_extensions = 6;
// TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
// used, we also need a map between RTP payload type and RTX payload type.
}
message VideoSendStreamConfig {
optional int64 timestamp_ms = 1;
// Synchronization source (stream identifier) for outgoing stream.
// One stream can have several ssrcs for e.g. simulcast.
optional uint32 ssrc = 2;
// SSRC for the RTX stream
optional uint32 rtx_ssrc = 3;
// RTP source stream ID
optional bytes rsid = 4;
// IDs for the header extension we care about. Only required if there are
// header extensions configured.
optional RtpHeaderExtensionConfig header_extensions = 5;
// TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
// used, we also need a map between RTP payload type and RTX payload type.
}
message AudioRecvStreamConfig {
optional int64 timestamp_ms = 1;
// required - Synchronization source (stream identifier) to be received.
optional uint32 remote_ssrc = 2;
// required - Sender SSRC used for sending RTCP (such as receiver reports).
optional uint32 local_ssrc = 3;
// Field number 4 reserved for RTX SSRC.
// optional - RTP source stream ID
optional bytes rsid = 5;
// IDs for the header extension we care about. Only required if there are
// header extensions configured.
optional RtpHeaderExtensionConfig header_extensions = 6;
// TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
// used, we also need a map between RTP payload type and RTX payload type.
}
message AudioSendStreamConfig {
optional int64 timestamp_ms = 1;
// Synchronization source (stream identifier) for outgoing stream.
// One stream can have several ssrcs for e.g. simulcast.
optional uint32 ssrc = 2;
// Field number 3 reserved for RTX SSRC
// RTP source stream ID
optional bytes rsid = 4;
// IDs for the header extension we care about. Only required if there are
// header extensions configured.
optional RtpHeaderExtensionConfig header_extensions = 5;
// TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
// used, we also need a map between RTP payload type and RTX payload type.
}
message AudioNetworkAdaptations {
optional int64 timestamp_ms = 1;
// Bit rate that the audio encoder is operating at.
// TODO(terelius): Signed vs unsigned?
optional int32 bitrate_bps = 2;
// Frame length that each encoded audio packet consists of.
// TODO(terelius): Signed vs unsigned?
optional int32 frame_length_ms = 3;
// Packet loss fraction that the encoder's forward error correction (FEC) is
// optimized for.
optional float uplink_packet_loss_fraction = 4;
// Whether forward error correction (FEC) is turned on or off.
optional bool enable_fec = 5;
// Whether discontinuous transmission (DTX) is turned on or off.
optional bool enable_dtx = 6;
// Number of audio channels that each encoded packet consists of.
optional uint32 num_channels = 7;
// Delta encodings
optional bytes timestamp_deltas_ms = 101;
optional bytes bitrate_deltas_bps = 102;
optional bytes frame_length_deltas_ms = 103;
optional bytes uplink_packet_loss_fraction_deltas = 104;
optional bytes enable_fec_deltas = 105;
optional bytes enable_dtx_deltas = 106;
optional bytes num_channels_deltas = 107;
}
message BweProbeCluster {
optional int64 timestamp_ms = 1;
// required - The id of this probe cluster.
optional uint32 id = 2;
// required - The bitrate in bps that this probe cluster is meant to probe.
optional uint32 bitrate_bps = 3;
// required - The minimum number of packets used to probe the given bitrate.
optional uint32 min_packets = 4;
// required - The minimum number of bytes used to probe the given bitrate.
optional uint32 min_bytes = 5;
}
message BweProbeResultSuccess {
optional int64 timestamp_ms = 1;
// required - The id of this probe cluster.
optional uint32 id = 2;
// required - The resulting bitrate in bps.
optional uint32 bitrate_bps = 3;
}
message BweProbeResultFailure {
optional int64 timestamp_ms = 1;
// required - The id of this probe cluster.
optional uint32 id = 2;
enum FailureReason {
UNKNOWN = 0;
INVALID_SEND_RECEIVE_INTERVAL = 1;
INVALID_SEND_RECEIVE_RATIO = 2;
TIMEOUT = 3;
}
// required
optional FailureReason failure = 3;
}