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_TEST_CHANNEL_H_
12#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
13
14#include <stdio.h>
15
16#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
17#include "webrtc/modules/include/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  size_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  size_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  int32_t SendData(FrameType frameType,
54                   uint8_t payloadType,
55                   uint32_t timeStamp,
56                   const uint8_t* payloadData,
57                   size_t payloadSize,
58                   const RTPFragmentationHeader* fragmentation) override;
59
60  void RegisterReceiverACM(AudioCodingModule *acm);
61
62  void ResetStats();
63
64  int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
65
66  void Stats(uint32_t* numPackets);
67
68  void Stats(uint8_t* payloadType, uint32_t* payloadLenByte);
69
70  void PrintStats(CodecInst& codecInst);
71
72  void SetIsStereo(bool isStereo) {
73    _isStereo = isStereo;
74  }
75
76  uint32_t LastInTimestamp();
77
78  void SetFECTestWithPacketLoss(bool usePacketLoss) {
79    _useFECTestWithPacketLoss = usePacketLoss;
80  }
81
82  double BitRate();
83
84  void set_send_timestamp(uint32_t new_send_ts) {
85    external_send_timestamp_ = new_send_ts;
86  }
87
88  void set_sequence_number(uint16_t new_sequence_number) {
89    external_sequence_number_ = new_sequence_number;
90  }
91
92  void set_num_packets_to_drop(int new_num_packets_to_drop) {
93    num_packets_to_drop_ = new_num_packets_to_drop;
94  }
95
96 private:
97  void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
98
99  AudioCodingModule* _receiverACM;
100  uint16_t _seqNo;
101  // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
102  uint8_t _payloadData[60 * 32 * 2 * 2];
103
104  CriticalSectionWrapper* _channelCritSect;
105  FILE* _bitStreamFile;
106  bool _saveBitStream;
107  int16_t _lastPayloadType;
108  ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
109  bool _isStereo;
110  WebRtcRTPHeader _rtpInfo;
111  bool _leftChannel;
112  uint32_t _lastInTimestamp;
113  bool _useLastFrameSize;
114  uint32_t _lastFrameSizeSample;
115  // FEC Test variables
116  int16_t _packetLoss;
117  bool _useFECTestWithPacketLoss;
118  uint64_t _beginTime;
119  uint64_t _totalBytes;
120
121  // External timing info, defaulted to -1. Only used if they are
122  // non-negative.
123  int64_t external_send_timestamp_;
124  int32_t external_sequence_number_;
125  int num_packets_to_drop_;
126};
127
128}  // namespace webrtc
129
130#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
131