Reland "New protobuf format for event log.""""
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}diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index 981231b..5bf99da 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -137,6 +137,13 @@
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 @@
]
deps = [
+ ":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_proto",
"..:webrtc_common",
diff --git a/logging/rtc_event_log/rtc_event_log2.proto b/logging/rtc_event_log/rtc_event_log2.proto
new file mode 100644
index 0000000..4cac455
--- /dev/null
+++ b/logging/rtc_event_log/rtc_event_log2.proto
@@ -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;
+}
