Index: webrtc/modules/audio_coding/neteq/neteq_unittest.proto |
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.proto b/webrtc/modules/audio_coding/neteq/neteq_unittest.proto |
new file mode 100644 |
index 0000000000000000000000000000000000000000..4b59848eb26f3bc60cc519da9737e3f065fb9798 |
--- /dev/null |
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.proto |
@@ -0,0 +1,29 @@ |
+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; |
+} |