audio_coding: remove "main" directory

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

Index: webrtc/modules/audio_coding/main/test/PacketLossTest.cc
diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc b/webrtc/modules/audio_coding/main/test/PacketLossTest.cc
deleted file mode 100644
index f7c96faacd691e994e2768f1eb5c0a6cee501583..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc
+++ /dev/null
@@ -1,167 +0,0 @@
-/*
- *  Copyright (c) 2014 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.
- */
-
-#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-ReceiverWithPacketLoss::ReceiverWithPacketLoss()
-    : loss_rate_(0),
-      burst_length_(1),
-      packet_counter_(0),
-      lost_packet_counter_(0),
-      burst_lost_counter_(burst_length_) {
-}
-
-void ReceiverWithPacketLoss::Setup(AudioCodingModule *acm,
-                                   RTPStream *rtpStream,
-                                   std::string out_file_name,
-                                   int channels,
-                                   int loss_rate,
-                                   int burst_length) {
-  loss_rate_ = loss_rate;
-  burst_length_ = burst_length;
-  burst_lost_counter_ = burst_length_;  // To prevent first packet gets lost.
-  std::stringstream ss;
-  ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
-  Receiver::Setup(acm, rtpStream, ss.str(), channels);
-}
-
-bool ReceiverWithPacketLoss::IncomingPacket() {
-  if (!_rtpStream->EndOfFile()) {
-    if (packet_counter_ == 0) {
-      _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
-                                               _payloadSizeBytes, &_nextTime);
-      if (_realPayloadSizeBytes == 0) {
-        if (_rtpStream->EndOfFile()) {
-          packet_counter_ = 0;
-          return true;
-        } else {
-          return false;
-        }
-      }
-    }
-
-    if (!PacketLost()) {
-      _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
-    }
-    packet_counter_++;
-    _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
-                                             _payloadSizeBytes, &_nextTime);
-    if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
-      packet_counter_ = 0;
-      lost_packet_counter_ = 0;
-    }
-  }
-  return true;
-}
-
-bool ReceiverWithPacketLoss::PacketLost() {
-  if (burst_lost_counter_ < burst_length_) {
-    lost_packet_counter_++;
-    burst_lost_counter_++;
-    return true;
-  }
-
-  if (lost_packet_counter_ * 100 < loss_rate_ * packet_counter_) {
-    lost_packet_counter_++;
-    burst_lost_counter_ = 1;
-    return true;
-  }
-  return false;
-}
-
-SenderWithFEC::SenderWithFEC()
-    : expected_loss_rate_(0) {
-}
-
-void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                          std::string in_file_name, int sample_rate,
-                          int channels, int expected_loss_rate) {
-  Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
-  EXPECT_TRUE(SetFEC(true));
-  EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
-}
-
-bool SenderWithFEC::SetFEC(bool enable_fec) {
-  if (_acm->SetCodecFEC(enable_fec) == 0) {
-    return true;
-  }
-  return false;
-}
-
-bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
-  if (_acm->SetPacketLossRate(expected_loss_rate) == 0) {
-    expected_loss_rate_ = expected_loss_rate;
-    return true;
-  }
-  return false;
-}
-
-PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
-                               int actual_loss_rate, int burst_length)
-    : channels_(channels),
-      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
-                    "audio_coding/teststereo32kHz"),
-      sample_rate_hz_(32000),
-      sender_(new SenderWithFEC),
-      receiver_(new ReceiverWithPacketLoss),
-      expected_loss_rate_(expected_loss_rate),
-      actual_loss_rate_(actual_loss_rate),
-      burst_length_(burst_length) {
-}
-
-void PacketLossTest::Perform() {
-#ifndef WEBRTC_CODEC_OPUS
-  return;
-#else
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
-
-  int codec_id = acm->Codec("opus", 48000, channels_);
-
-  RTPFile rtpFile;
-  std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
-                                                    "packet_loss_test");
-
-  // Encode to file
-  rtpFile.Open(fileName.c_str(), "wb+");
-  rtpFile.WriteHeader();
-
-  sender_->testMode = 0;
-  sender_->codeId = codec_id;
-
-  sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
-                 expected_loss_rate_);
-  if (acm->SendCodec()) {
-    sender_->Run();
-  }
-  sender_->Teardown();
-  rtpFile.Close();
-
-  // Decode to file
-  rtpFile.Open(fileName.c_str(), "rb");
-  rtpFile.ReadHeader();
-
-  receiver_->testMode = 0;
-  receiver_->codeId = codec_id;
-
-  receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
-                   actual_loss_rate_, burst_length_);
-  receiver_->Run();
-  receiver_->Teardown();
-  rtpFile.Close();
-#endif
-}
-
-}  // namespace webrtc