1/*
2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
12#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
13
14#include <stdio.h>
15
16#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
17#include "webrtc/modules/interface/module_common_types.h"
18#include "webrtc/typedefs.h"
19
20namespace webrtc {
21
22class CriticalSectionWrapper;
23
24#define MAX_NUM_PAYLOADS   50
25#define MAX_NUM_FRAMESIZES  6
26
27// TODO(turajs): Write constructor for this structure.
28struct ACMTestFrameSizeStats {
29  uint16_t frameSizeSample;
30  int16_t maxPayloadLen;
31  uint32_t numPackets;
32  uint64_t totalPayloadLenByte;
33  uint64_t totalEncodedSamples;
34  double rateBitPerSec;
35  double usageLenSec;
36};
37
38// TODO(turajs): Write constructor for this structure.
39struct ACMTestPayloadStats {
40  bool newPacket;
41  int16_t payloadType;
42  int16_t lastPayloadLenByte;
43  uint32_t lastTimestamp;
44  ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
45};
46
47class Channel : public AudioPacketizationCallback {
48 public:
49
50  Channel(int16_t chID = -1);
51  ~Channel();
52
53  virtual int32_t SendData(
54      const FrameType frameType, const uint8_t payloadType,
55      const uint32_t timeStamp, const uint8_t* payloadData,
56      const uint16_t payloadSize,
57      const RTPFragmentationHeader* fragmentation) OVERRIDE;
58
59  void RegisterReceiverACM(AudioCodingModule *acm);
60
61  void ResetStats();
62
63  int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
64
65  void Stats(uint32_t* numPackets);
66
67  void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
68
69  void PrintStats(CodecInst& codecInst);
70
71  void SetIsStereo(bool isStereo) {
72    _isStereo = isStereo;
73  }
74
75  uint32_t LastInTimestamp();
76
77  void SetFECTestWithPacketLoss(bool usePacketLoss) {
78    _useFECTestWithPacketLoss = usePacketLoss;
79  }
80
81  double BitRate();
82
83  void set_send_timestamp(uint32_t new_send_ts) {
84    external_send_timestamp_ = new_send_ts;
85  }
86
87  void set_sequence_number(uint16_t new_sequence_number) {
88    external_sequence_number_ = new_sequence_number;
89  }
90
91  void set_num_packets_to_drop(int new_num_packets_to_drop) {
92    num_packets_to_drop_ = new_num_packets_to_drop;
93  }
94
95 private:
96  void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
97
98  AudioCodingModule* _receiverACM;
99  uint16_t _seqNo;
100  // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
101  uint8_t _payloadData[60 * 32 * 2 * 2];
102
103  CriticalSectionWrapper* _channelCritSect;
104  FILE* _bitStreamFile;
105  bool _saveBitStream;
106  int16_t _lastPayloadType;
107  ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
108  bool _isStereo;
109  WebRtcRTPHeader _rtpInfo;
110  bool _leftChannel;
111  uint32_t _lastInTimestamp;
112  // FEC Test variables
113  int16_t _packetLoss;
114  bool _useFECTestWithPacketLoss;
115  uint64_t _beginTime;
116  uint64_t _totalBytes;
117
118  // External timing info, defaulted to -1. Only used if they are
119  // non-negative.
120  int64_t external_send_timestamp_;
121  int32_t external_sequence_number_;
122  int num_packets_to_drop_;
123};
124
125}  // namespace webrtc
126
127#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
128