/*
 *  Copyright (c) 2011 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 CHANNEL_H
#define CHANNEL_H

#include <stdio.h>

#include "audio_coding_module.h"
#include "critical_section_wrapper.h"
#include "rw_lock_wrapper.h"


#define MAX_NUM_PAYLOADS   50
#define MAX_NUM_FRAMESIZES  6


struct ACMTestFrameSizeStats
{
    WebRtc_UWord16 frameSizeSample;
    WebRtc_Word16  maxPayloadLen;
    WebRtc_UWord32 numPackets;
    WebRtc_UWord64 totalPayloadLenByte;
    WebRtc_UWord64 totalEncodedSamples;
    double       rateBitPerSec;
    double       usageLenSec;
           
};

struct ACMTestPayloadStats
{
    bool                  newPacket;
    WebRtc_Word16           payloadType;
    WebRtc_Word16           lastPayloadLenByte;
    WebRtc_UWord32          lastTimestamp;
    ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
};

using namespace webrtc;

class Channel: public AudioPacketizationCallback
{
public:

    Channel(
        WebRtc_Word16 chID = -1);
    ~Channel();

    WebRtc_Word32 SendData(
        const FrameType       frameType,
        const WebRtc_UWord8   payloadType,
        const WebRtc_UWord32  timeStamp,
        const WebRtc_UWord8*  payloadData, 
        const WebRtc_UWord16  payloadSize,
        const RTPFragmentationHeader* fragmentation);

    void RegisterReceiverACM(
        AudioCodingModule *acm);
    
    void ResetStats();
    
    WebRtc_Word16 Stats(
        CodecInst&     codecInst, 
        ACMTestPayloadStats& payloadStats);
    
    void Stats(
        WebRtc_UWord32* numPackets);
    
    void Stats(
        WebRtc_UWord8*  payloadLenByte, 
        WebRtc_UWord32* payloadType);
    
    void PrintStats(
        CodecInst& codecInst);
    
    void SetIsStereo(bool isStereo)
    {
        _isStereo = isStereo;
    }

    WebRtc_UWord32 LastInTimestamp();
    
    void SetFECTestWithPacketLoss(bool usePacketLoss)
    {
        _useFECTestWithPacketLoss = usePacketLoss;
    }

    double BitRate();

private:
    void CalcStatistics(
        WebRtcRTPHeader& rtpInfo, 
        WebRtc_UWord16   payloadSize);

    AudioCodingModule* _receiverACM;
    WebRtc_UWord16           _seqNo;
    // 60 msec * 32 sample (max) / msec * 2 description (maybe) * 2 bytes / sample
    WebRtc_UWord8            _payloadData[60 * 32 * 2 * 2]; 

    CriticalSectionWrapper* _channelCritSect;
    FILE*                _bitStreamFile;
    bool                 _saveBitStream;
    WebRtc_Word16          _lastPayloadType;
    ACMTestPayloadStats  _payloadStats[MAX_NUM_PAYLOADS];
    bool                 _isStereo;
    WebRtcRTPHeader        _rtpInfo;
    bool                 _leftChannel;
    WebRtc_UWord32         _lastInTimestamp;
    // FEC Test variables
    WebRtc_Word16          _packetLoss;
    bool                 _useFECTestWithPacketLoss;
    WebRtc_Word16          _chID;
    WebRtc_UWord64         _beginTime;
    WebRtc_UWord64         _totalBytes;
};


#endif