audio_coding: remove "main" directory

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

-/*
- *  Copyright (c) 2012 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_CHANNEL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
-
-#include <stdio.h>
-
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-
-#define MAX_NUM_PAYLOADS   50
-#define MAX_NUM_FRAMESIZES  6
-
-// TODO(turajs): Write constructor for this structure.
-struct ACMTestFrameSizeStats {
-  uint16_t frameSizeSample;
-  size_t maxPayloadLen;
-  uint32_t numPackets;
-  uint64_t totalPayloadLenByte;
-  uint64_t totalEncodedSamples;
-  double rateBitPerSec;
-  double usageLenSec;
-};
-
-// TODO(turajs): Write constructor for this structure.
-struct ACMTestPayloadStats {
-  bool newPacket;
-  int16_t payloadType;
-  size_t lastPayloadLenByte;
-  uint32_t lastTimestamp;
-  ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
-};
-
-class Channel : public AudioPacketizationCallback {
- public:
-
-  Channel(int16_t chID = -1);
-  ~Channel();
-
-  int32_t SendData(FrameType frameType,
-                   uint8_t payloadType,
-                   uint32_t timeStamp,
-                   const uint8_t* payloadData,
-                   size_t payloadSize,
-                   const RTPFragmentationHeader* fragmentation) override;
-
-  void RegisterReceiverACM(AudioCodingModule *acm);
-
-  void ResetStats();
-
-  int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
-
-  void Stats(uint32_t* numPackets);
-
-  void Stats(uint8_t* payloadType, uint32_t* payloadLenByte);
-
-  void PrintStats(CodecInst& codecInst);
-
-  void SetIsStereo(bool isStereo) {
-    _isStereo = isStereo;
-  }
-
-  uint32_t LastInTimestamp();
-
-  void SetFECTestWithPacketLoss(bool usePacketLoss) {
-    _useFECTestWithPacketLoss = usePacketLoss;
-  }
-
-  double BitRate();
-
-  void set_send_timestamp(uint32_t new_send_ts) {
-    external_send_timestamp_ = new_send_ts;
-  }
-
-  void set_sequence_number(uint16_t new_sequence_number) {
-    external_sequence_number_ = new_sequence_number;
-  }
-
-  void set_num_packets_to_drop(int new_num_packets_to_drop) {
-    num_packets_to_drop_ = new_num_packets_to_drop;
-  }
-
- private:
-  void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
-
-  AudioCodingModule* _receiverACM;
-  uint16_t _seqNo;
-  // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
-  uint8_t _payloadData[60 * 32 * 2 * 2];
-
-  CriticalSectionWrapper* _channelCritSect;
-  FILE* _bitStreamFile;
-  bool _saveBitStream;
-  int16_t _lastPayloadType;
-  ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
-  bool _isStereo;
-  WebRtcRTPHeader _rtpInfo;
-  bool _leftChannel;
-  uint32_t _lastInTimestamp;
-  bool _useLastFrameSize;
-  uint32_t _lastFrameSizeSample;
-  // FEC Test variables
-  int16_t _packetLoss;
-  bool _useFECTestWithPacketLoss;
-  uint64_t _beginTime;
-  uint64_t _totalBytes;
-
-  // External timing info, defaulted to -1. Only used if they are
-  // non-negative.
-  int64_t external_send_timestamp_;
-  int32_t external_sequence_number_;
-  int num_packets_to_drop_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_