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Side by Side Diff: webrtc/modules/audio_coding/test/Channel.h

Issue 1613643004: Remove mutable from rtc::CriticalSection members. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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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
(...skipping 82 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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