Update NetEq network statistics in neteq_unittest.

webrtc/modules/audio_coding/neteq/neteq_unittest.proto

+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;
+}