| 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
|
|
|