Delete a chain of methods in ViE, VoE and ACM

webrtc/modules/audio_coding/main/test/delay_test.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 15 matching lines @@
 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
 #include "webrtc/modules/audio_coding/main/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 DEFINE_string(codec, "isac", "Codec Name");
 DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
 DEFINE_int32(num_channels, 1, "Number of Channels.");
 DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
 DEFINE_int32(delay, 0, "Delay in millisecond.");
-DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
 
 namespace webrtc {
 
 namespace {
 
 struct CodecSettings {
   char name[50];
@@ skipping 35 matching lines @@
     test_cntr_ = 0;
     std::string file_name = webrtc::test::ResourcePath(
         "audio_coding/testfile32kHz", "pcm");
     if (FLAGS_input_file.size() > 0)
       file_name = FLAGS_input_file;
     in_file_a_.Open(file_name, 32000, "rb");
     ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
         "Couldn't initialize receiver.\n";
     ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
         "Couldn't initialize receiver.\n";
-    if (FLAGS_init_delay > 0) {
-      ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
-          "Failed to set initial delay.\n";
-    }
 
     if (FLAGS_delay > 0) {
       ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
           "Failed to set minimum delay.\n";
     }
 
     int num_encoders = acm_a_->NumberOfCodecs();
     CodecInst my_codec_param;
     for (int n = 0; n < num_encoders; n++) {
       EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
@@ skipping 59 matching lines @@
         "Failed to set RED.\n";
   }
 
   void ConfigChannel(bool packet_loss) {
     channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
   }
 
   void OpenOutFile(const char* output_id) {
     std::stringstream file_stream;
     file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
-        << "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
+        << "Hz" << "_" << FLAGS_delay << "ms.pcm";
     std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
     std::string file_name = webrtc::test::OutputPath() + file_stream.str();
     out_file_b_.Open(file_name.c_str(), 32000, "wb");
   }
 
   void Run(int duration_sec, const char* output_prefix) {
     OpenOutFile(output_prefix);
     AudioFrame audio_frame;
     uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
 
@@ skipping 75 matching lines @@
   test_setting.codec.num_channels = FLAGS_num_channels;
   test_setting.acm.dtx = FLAGS_dtx;
   test_setting.acm.fec = FLAGS_fec;
   test_setting.packet_loss = FLAGS_packet_loss;
 
   webrtc::DelayTest delay_test;
   delay_test.Initialize();
   delay_test.Perform(&test_setting, 1, 240, "delay_test");
   return 0;
 }