webrtc/modules/audio_coding/neteq/neteq_unittest.proto


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syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.neteq_unittest;

message NetEqNetworkStatistics {
  optional uint32 current_buffer_size_ms = 1;
  optional uint32 preferred_buffer_size_ms = 2;
  optional uint32 jitter_peaks_found = 3;
  optional uint32 packet_loss_rate = 4;
  optional uint32 packet_discard_rate = 5;
  optional uint32 expand_rate = 6;
  optional uint32 speech_expand_rate = 7;
  optional uint32 preemptive_rate = 8;
  optional uint32 accelerate_rate = 9;
  optional uint32 secondary_decoded_rate = 10;
  optional int32 clockdrift_ppm = 11;
  optional uint64 added_zero_samples = 12;
  optional int32 mean_waiting_time_ms = 13;
  optional int32 median_waiting_time_ms = 14;
  optional int32 min_waiting_time_ms = 15;
  optional int32 max_waiting_time_ms = 16;
}

message RtcpStatistics {
  optional uint32 fraction_lost = 1;
  optional uint32 cumulative_lost = 2;
  optional uint32 extended_max_sequence_number = 3;
  optional uint32 jitter = 4;
}