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/external/webrtc/webrtc/modules/audio_coding/test/
DChannel.cc139 if (_lastPayloadType == _payloadStats[n].payloadType) { in CalcStatistics()
140 _payloadStats[n].newPacket = true; in CalcStatistics()
150 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics()
152 currentPayloadStr = &_payloadStats[n]; in CalcStatistics()
212 while (_payloadStats[n].payloadType != -1) { in CalcStatistics()
216 _payloadStats[n].newPacket = false; in CalcStatistics()
217 _payloadStats[n].lastPayloadLenByte = payloadSize; in CalcStatistics()
218 _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp; in CalcStatistics()
219 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics()
220 memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES * in CalcStatistics()
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DChannel.h108 ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS]; variable