| Index: webrtc/video/rtc_event_log.proto
 | 
| diff --git a/webrtc/video/rtc_event_log.proto b/webrtc/video/rtc_event_log.proto
 | 
| deleted file mode 100644
 | 
| index 7e4e699e338c1bcbaf44e397942d600d2dbd11a4..0000000000000000000000000000000000000000
 | 
| --- a/webrtc/video/rtc_event_log.proto
 | 
| +++ /dev/null
 | 
| @@ -1,228 +0,0 @@
 | 
| -syntax = "proto2";
 | 
| -option optimize_for = LITE_RUNTIME;
 | 
| -package webrtc.rtclog;
 | 
| -
 | 
| -
 | 
| -enum MediaType {
 | 
| -  ANY = 0;
 | 
| -  AUDIO = 1;
 | 
| -  VIDEO = 2;
 | 
| -  DATA = 3;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -// This is the main message to dump to a file, it can contain multiple event
 | 
| -// messages, but it is possible to append multiple EventStreams (each with a
 | 
| -// single event) to a file.
 | 
| -// This has the benefit that there's no need to keep all data in memory.
 | 
| -message EventStream {
 | 
| -  repeated Event stream = 1;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message Event {
 | 
| -  // required - Elapsed wallclock time in us since the start of the log.
 | 
| -  optional int64 timestamp_us = 1;
 | 
| -
 | 
| -  // The different types of events that can occur, the UNKNOWN_EVENT entry
 | 
| -  // is added in case future EventTypes are added, in that case old code will
 | 
| -  // receive the new events as UNKNOWN_EVENT.
 | 
| -  enum EventType {
 | 
| -    UNKNOWN_EVENT = 0;
 | 
| -    RTP_EVENT = 1;
 | 
| -    RTCP_EVENT = 2;
 | 
| -    DEBUG_EVENT = 3;
 | 
| -    VIDEO_RECEIVER_CONFIG_EVENT = 4;
 | 
| -    VIDEO_SENDER_CONFIG_EVENT = 5;
 | 
| -    AUDIO_RECEIVER_CONFIG_EVENT = 6;
 | 
| -    AUDIO_SENDER_CONFIG_EVENT = 7;
 | 
| -  }
 | 
| -
 | 
| -  // required - Indicates the type of this event
 | 
| -  optional EventType type = 2;
 | 
| -
 | 
| -  // optional - but required if type == RTP_EVENT
 | 
| -  optional RtpPacket rtp_packet = 3;
 | 
| -
 | 
| -  // optional - but required if type == RTCP_EVENT
 | 
| -  optional RtcpPacket rtcp_packet = 4;
 | 
| -
 | 
| -  // optional - but required if type == DEBUG_EVENT
 | 
| -  optional DebugEvent debug_event = 5;
 | 
| -
 | 
| -  // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
 | 
| -  optional VideoReceiveConfig video_receiver_config = 6;
 | 
| -
 | 
| -  // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
 | 
| -  optional VideoSendConfig video_sender_config = 7;
 | 
| -
 | 
| -  // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
 | 
| -  optional AudioReceiveConfig audio_receiver_config = 8;
 | 
| -
 | 
| -  // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
 | 
| -  optional AudioSendConfig audio_sender_config = 9;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message RtpPacket {
 | 
| -  // required - True if the packet is incoming w.r.t. the user logging the data
 | 
| -  optional bool incoming = 1;
 | 
| -
 | 
| -  // required
 | 
| -  optional MediaType type = 2;
 | 
| -
 | 
| -  // required - The size of the packet including both payload and header.
 | 
| -  optional uint32 packet_length = 3;
 | 
| -
 | 
| -  // required - The RTP header only.
 | 
| -  optional bytes header = 4;
 | 
| -
 | 
| -  // Do not add code to log user payload data without a privacy review!
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message RtcpPacket {
 | 
| -  // required - True if the packet is incoming w.r.t. the user logging the data
 | 
| -  optional bool incoming = 1;
 | 
| -
 | 
| -  // required
 | 
| -  optional MediaType type = 2;
 | 
| -
 | 
| -  // required - The whole packet including both payload and header.
 | 
| -  optional bytes packet_data = 3;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message DebugEvent {
 | 
| -  // Indicates the type of the debug event.
 | 
| -  // LOG_START and LOG_END indicate the start and end of the log respectively.
 | 
| -  // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM.
 | 
| -  enum EventType {
 | 
| -    UNKNOWN_EVENT = 0;
 | 
| -    LOG_START = 1;
 | 
| -    LOG_END = 2;
 | 
| -    AUDIO_PLAYOUT = 3;
 | 
| -  }
 | 
| -
 | 
| -  // required
 | 
| -  optional EventType type = 1;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -// TODO(terelius): Video and audio streams could in principle share SSRC,
 | 
| -// so identifying a stream based only on SSRC might not work.
 | 
| -// It might be better to use a combination of SSRC and media type
 | 
| -// or SSRC and port number, but for now we will rely on SSRC only.
 | 
| -message VideoReceiveConfig {
 | 
| -  // required - Synchronization source (stream identifier) to be received.
 | 
| -  optional uint32 remote_ssrc = 1;
 | 
| -  // required - Sender SSRC used for sending RTCP (such as receiver reports).
 | 
| -  optional uint32 local_ssrc = 2;
 | 
| -
 | 
| -  // Compound mode is described by RFC 4585 and reduced-size
 | 
| -  // RTCP mode is described by RFC 5506.
 | 
| -  enum RtcpMode {
 | 
| -    RTCP_COMPOUND = 1;
 | 
| -    RTCP_REDUCEDSIZE = 2;
 | 
| -  }
 | 
| -  // required - RTCP mode to use.
 | 
| -  optional RtcpMode rtcp_mode = 3;
 | 
| -
 | 
| -  // required - Extended RTCP settings.
 | 
| -  optional bool receiver_reference_time_report = 4;
 | 
| -
 | 
| -  // required - Receiver estimated maximum bandwidth.
 | 
| -  optional bool remb = 5;
 | 
| -
 | 
| -  // Map from video RTP payload type -> RTX config.
 | 
| -  repeated RtxMap rtx_map = 6;
 | 
| -
 | 
| -  // RTP header extensions used for the received stream.
 | 
| -  repeated RtpHeaderExtension header_extensions = 7;
 | 
| -
 | 
| -  // List of decoders associated with the stream.
 | 
| -  repeated DecoderConfig decoders = 8;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -// Maps decoder names to payload types.
 | 
| -message DecoderConfig {
 | 
| -  // required
 | 
| -  optional string name = 1;
 | 
| -
 | 
| -  // required
 | 
| -  optional sint32 payload_type = 2;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -// Maps RTP header extension names to numerical IDs.
 | 
| -message RtpHeaderExtension {
 | 
| -  // required
 | 
| -  optional string name = 1;
 | 
| -
 | 
| -  // required
 | 
| -  optional sint32 id = 2;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -// RTX settings for incoming video payloads that may be received.
 | 
| -// RTX is disabled if there's no config present.
 | 
| -message RtxConfig {
 | 
| -  // required - SSRC to use for the RTX stream.
 | 
| -  optional uint32 rtx_ssrc = 1;
 | 
| -
 | 
| -  // required - Payload type to use for the RTX stream.
 | 
| -  optional sint32 rtx_payload_type = 2;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message RtxMap {
 | 
| -  // required
 | 
| -  optional sint32 payload_type = 1;
 | 
| -
 | 
| -  // required
 | 
| -  optional RtxConfig config = 2;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message VideoSendConfig {
 | 
| -  // Synchronization source (stream identifier) for outgoing stream.
 | 
| -  // One stream can have several ssrcs for e.g. simulcast.
 | 
| -  // At least one ssrc is required.
 | 
| -  repeated uint32 ssrcs = 1;
 | 
| -
 | 
| -  // RTP header extensions used for the outgoing stream.
 | 
| -  repeated RtpHeaderExtension header_extensions = 2;
 | 
| -
 | 
| -  // List of SSRCs for retransmitted packets.
 | 
| -  repeated uint32 rtx_ssrcs = 3;
 | 
| -
 | 
| -  // required if rtx_ssrcs is used - Payload type for retransmitted packets.
 | 
| -  optional sint32 rtx_payload_type = 4;
 | 
| -
 | 
| -  // required - Canonical end-point identifier.
 | 
| -  optional string c_name = 5;
 | 
| -
 | 
| -  // required - Encoder associated with the stream.
 | 
| -  optional EncoderConfig encoder = 6;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -// Maps encoder names to payload types.
 | 
| -message EncoderConfig {
 | 
| -  // required
 | 
| -  optional string name = 1;
 | 
| -
 | 
| -  // required
 | 
| -  optional sint32 payload_type = 2;
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message AudioReceiveConfig {
 | 
| -  // TODO(terelius): Add audio-receive config.
 | 
| -}
 | 
| -
 | 
| -
 | 
| -message AudioSendConfig {
 | 
| -  // TODO(terelius): Add audio-receive config.
 | 
| -}
 | 
| 
 |