audio_coding: remove "main" directory

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

-/*
- *  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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
-
-#include <string>
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
-
-namespace webrtc {
-
-class ReceiverWithPacketLoss : public Receiver {
- public:
-  ReceiverWithPacketLoss();
-  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string out_file_name, int channels, int loss_rate,
-             int burst_length);
-  bool IncomingPacket() override;
-
- protected:
-  bool PacketLost();
-  int loss_rate_;
-  int burst_length_;
-  int packet_counter_;
-  int lost_packet_counter_;
-  int burst_lost_counter_;
-};
-
-class SenderWithFEC : public Sender {
- public:
-  SenderWithFEC();
-  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string in_file_name, int sample_rate, int channels,
-             int expected_loss_rate);
-  bool SetPacketLossRate(int expected_loss_rate);
-  bool SetFEC(bool enable_fec);
- protected:
-  int expected_loss_rate_;
-};
-
-class PacketLossTest : public ACMTest {
- public:
-  PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate,
-                 int burst_length);
-  void Perform();
- protected:
-  int channels_;
-  std::string in_file_name_;
-  int sample_rate_hz_;
-  rtc::scoped_ptr<SenderWithFEC> sender_;
-  rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_;
-  int expected_loss_rate_;
-  int actual_loss_rate_;
-  int burst_length_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_