Index: webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
index ef0e31e2f6a0ded47c0aa494becebb5a6f5ea714..8d0cf90d178e6ae2347eb202522a9cf6b1e4cfd7 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
@@ -80,7 +80,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, |
AudioFrame out_frame; |
while (time_now_ms < runtime_ms) { |
while (packet_input_time_ms <= time_now_ms) { |
- // Drop every N packets, where N = FLAGS_lossrate. |
+ // Drop every N packets, where N = FLAG_lossrate. |
bool lost = false; |
if (lossrate > 0) { |
lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0; |