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source: webrtc/webrtc/modules/audio_coding/main/test/Channel.h @ 0:4bda6873e34c

pub_scrub_3792 tip
Last change on this file since 0:4bda6873e34c was 0:4bda6873e34c, checked in by Michael Prittie <mprittie@…>, 6 years ago

Scrubbed password for publication.

File size: 3.3 KB
Line 
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
27struct ACMTestFrameSizeStats {
28  uint16_t frameSizeSample;
29  int16_t maxPayloadLen;
30  uint32_t numPackets;
31  uint64_t totalPayloadLenByte;
32  uint64_t totalEncodedSamples;
33  double rateBitPerSec;
34  double usageLenSec;
35};
36
37struct ACMTestPayloadStats {
38  bool newPacket;
39  int16_t payloadType;
40  int16_t lastPayloadLenByte;
41  uint32_t lastTimestamp;
42  ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
43};
44
45class Channel : public AudioPacketizationCallback {
46 public:
47
48  Channel(int16_t chID = -1);
49  ~Channel();
50
51  int32_t SendData(const FrameType frameType, const uint8_t payloadType,
52                   const uint32_t timeStamp, const uint8_t* payloadData,
53                   const uint16_t payloadSize,
54                   const RTPFragmentationHeader* fragmentation);
55
56  void RegisterReceiverACM(AudioCodingModule *acm);
57
58  void ResetStats();
59
60  int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
61
62  void Stats(uint32_t* numPackets);
63
64  void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
65
66  void PrintStats(CodecInst& codecInst);
67
68  void SetIsStereo(bool isStereo) {
69    _isStereo = isStereo;
70  }
71
72  uint32_t LastInTimestamp();
73
74  void SetFECTestWithPacketLoss(bool usePacketLoss) {
75    _useFECTestWithPacketLoss = usePacketLoss;
76  }
77
78  double BitRate();
79
80  void set_send_timestamp(uint32_t new_send_ts) {
81    external_send_timestamp_ = new_send_ts;
82  }
83
84  void set_sequence_number(uint16_t new_sequence_number) {
85    external_sequence_number_ = new_sequence_number;
86  }
87
88  void set_num_packets_to_drop(int new_num_packets_to_drop) {
89    num_packets_to_drop_ = new_num_packets_to_drop;
90  }
91
92 private:
93  void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
94
95  AudioCodingModule* _receiverACM;
96  uint16_t _seqNo;
97  // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
98  uint8_t _payloadData[60 * 32 * 2 * 2];
99
100  CriticalSectionWrapper* _channelCritSect;
101  FILE* _bitStreamFile;
102  bool _saveBitStream;
103  int16_t _lastPayloadType;
104  ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
105  bool _isStereo;
106  WebRtcRTPHeader _rtpInfo;
107  bool _leftChannel;
108  uint32_t _lastInTimestamp;
109  // FEC Test variables
110  int16_t _packetLoss;
111  bool _useFECTestWithPacketLoss;
112  uint64_t _beginTime;
113  uint64_t _totalBytes;
114
115  // External timing info, defaulted to -1. Only used if they are
116  // non-negative.
117  int64_t external_send_timestamp_;
118  int32_t external_sequence_number_;
119  int num_packets_to_drop_;
120};
121
122}  // namespace webrtc
123
124#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
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