logging/rtc_event_log/rtc_event_log2.proto - src - Git at Google

 // 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;
 }
	// 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;
	}