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  /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
rtp_format_h264.h 55 bool last_fragment,
61 last_fragment(last_fragment),
68 bool last_fragment; member in struct:webrtc::RtpPacketizerH264::Packet
rtp_format_h264.cc 194 packets_.back().last_fragment = true;
210 if (packet.first_fragment && packet.last_fragment) {
245 if (packet.last_fragment)
249 assert(packet.last_fragment);
263 fu_header |= (packet.last_fragment ? kEBit : 0);
269 if (packet.last_fragment) {

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