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 32c085d3e0be027ae5c8d443e6d389a05f0dec94..59402a2029b3ab2e3ff49f27312a7ddf24382f3f 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc |
@@ -75,6 +75,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, |
// Main loop. |
webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock(); |
int64_t start_time_ms = clock->TimeInMilliseconds(); |
+ 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. |
@@ -104,7 +105,6 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, |
} |
// Get output audio, but don't do anything with it. |
- AudioFrame out_frame; |
int error = neteq->GetAudio(&out_frame); |
if (error != NetEq::kOK) |
return -1; |