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  /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
insert_packet_with_timing.cc 72 loss_threshold_(0),
129 loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
131 loss_threshold_ = 0;
177 if (loss_threshold_ > 0 && rand() < loss_threshold_) {
275 double loss_threshold_; member in class:webrtc::InsertPacketWithTiming

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