Index: webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc |
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc |
index ea7266567e442ce532683275e696474fada8521a..5b1e07e47801948c6af4210891345e7f6faf914a 100644 |
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc |
+++ b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc |
@@ -42,7 +42,6 @@ DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file"); |
DEFINE_string(delay, "", "Log for delay."); |
// Other setups |
-DEFINE_int32(init_delay, 0, "Initial delay."); |
DEFINE_bool(verbose, false, "Verbosity."); |
DEFINE_double(loss_rate, 0, "Rate of packet loss < 1"); |
@@ -122,9 +121,6 @@ class InsertPacketWithTiming { |
<< " Hz." << std::endl; |
// Other setups |
- if (FLAGS_init_delay > 0) |
- EXPECT_EQ(0, receive_acm_->SetInitialPlayoutDelay(FLAGS_init_delay)); |
- |
if (FLAGS_loss_rate > 0) |
loss_threshold_ = RAND_MAX * FLAGS_loss_rate; |
else |