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  /external/webrtc/webrtc/modules/audio_coding/test/
Channel.cc 139 if (_lastPayloadType == _payloadStats[n].payloadType) {
140 _payloadStats[n].newPacket = true;
150 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
152 currentPayloadStr = &_payloadStats[n];
212 while (_payloadStats[n].payloadType != -1) {
216 _payloadStats[n].newPacket = false;
217 _payloadStats[n].lastPayloadLenByte = payloadSize;
218 _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
219 _payloadStats[n].payloadType = rtpInfo.header.payloadType;
220 memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES
    [all...]
Channel.h 108 ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];

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