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Unified Diff: webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc

Issue 1421013006: Delete a chain of methods in ViE, VoE and ACM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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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
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