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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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93 } | 93 } |
94 | 94 |
95 private: | 95 private: |
96 void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize); | 96 void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize); |
97 | 97 |
98 AudioCodingModule* _receiverACM; | 98 AudioCodingModule* _receiverACM; |
99 uint16_t _seqNo; | 99 uint16_t _seqNo; |
100 // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample | 100 // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample |
101 uint8_t _payloadData[60 * 32 * 2 * 2]; | 101 uint8_t _payloadData[60 * 32 * 2 * 2]; |
102 | 102 |
103 mutable rtc::CriticalSection _channelCritSect; | 103 rtc::CriticalSection _channelCritSect; |
104 FILE* _bitStreamFile; | 104 FILE* _bitStreamFile; |
105 bool _saveBitStream; | 105 bool _saveBitStream; |
106 int16_t _lastPayloadType; | 106 int16_t _lastPayloadType; |
107 ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS]; | 107 ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS]; |
108 bool _isStereo; | 108 bool _isStereo; |
109 WebRtcRTPHeader _rtpInfo; | 109 WebRtcRTPHeader _rtpInfo; |
110 bool _leftChannel; | 110 bool _leftChannel; |
111 uint32_t _lastInTimestamp; | 111 uint32_t _lastInTimestamp; |
112 bool _useLastFrameSize; | 112 bool _useLastFrameSize; |
113 uint32_t _lastFrameSizeSample; | 113 uint32_t _lastFrameSizeSample; |
114 // FEC Test variables | 114 // FEC Test variables |
115 int16_t _packetLoss; | 115 int16_t _packetLoss; |
116 bool _useFECTestWithPacketLoss; | 116 bool _useFECTestWithPacketLoss; |
117 uint64_t _beginTime; | 117 uint64_t _beginTime; |
118 uint64_t _totalBytes; | 118 uint64_t _totalBytes; |
119 | 119 |
120 // External timing info, defaulted to -1. Only used if they are | 120 // External timing info, defaulted to -1. Only used if they are |
121 // non-negative. | 121 // non-negative. |
122 int64_t external_send_timestamp_; | 122 int64_t external_send_timestamp_; |
123 int32_t external_sequence_number_; | 123 int32_t external_sequence_number_; |
124 int num_packets_to_drop_; | 124 int num_packets_to_drop_; |
125 }; | 125 }; |
126 | 126 |
127 } // namespace webrtc | 127 } // namespace webrtc |
128 | 128 |
129 #endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ | 129 #endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ |
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