text: /*
   *  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/interface/audio_coding_module.h"
  #include "webrtc/modules/interface/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;
    int16_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;
    int16_t lastPayloadLenByte;
    uint32_t lastTimestamp;
    ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
  };
  
  class Channel : public AudioPacketizationCallback {
   public:
  
    Channel(int16_t chID = -1);
    ~Channel();
  
    virtual int32_t SendData(
        const FrameType frameType, const uint8_t payloadType,
        const uint32_t timeStamp, const uint8_t* payloadData,
        const uint16_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* payloadLenByte, uint32_t* payloadType);
  
    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, uint16_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;
    // 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_
  