| 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;
|
|
|