| /external/webrtc/webrtc/modules/video_coding/test/ |
| stream_generator.cc | 25 : packets_(), sequence_number_(start_seq_num), start_time_(current_time) {}
28 packets_.clear();
43 packets_.push_back(GeneratePacket(sequence_number_, timestamp, packet_size,
48 packets_.push_back(GeneratePacket(sequence_number_, timestamp, 0, false,
80 if (it == packets_.end())
84 packets_.erase(it);
90 if (it == packets_.end())
98 if (packets_.empty())
101 *packet = packets_.front();
102 packets_.pop_front()
[all...] |
| rtp_player.cc | 80 packets_(),
91 while (!packets_.empty()) {
92 delete packets_.back();
93 packets_.pop_back();
105 packets_.push_back(packet);
113 for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
130 for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
134 packets_.erase(it)
183 RtpPacketList packets_; member in class:webrtc::rtpplayer::LostPackets
[all...] |
| stream_generator.h | 62 std::list<VCMPacket> packets_; member in class:webrtc::StreamGenerator
|
| /external/webrtc/webrtc/modules/video_coding/ |
| session_info.cc | 31 packets_(),
39 for (PacketIterator it = packets_.begin(); it != packets_.end(); ++it)
47 if (packets_.empty())
49 return packets_.front().seqNum;
53 if (packets_.empty())
56 return packets_.back().seqNum;
57 return LatestSequenceNumber(packets_.back().seqNum, empty_seq_num_high_);
61 if (packets_.empty())
63 if (packets_.front().codecSpecificHeader.codec == kRtpVideoVp8)
[all...] |
| session_info.h | 108 // none is found the returned iterator points to |packets_.end()|.
155 PacketList packets_; member in class:webrtc::VCMSessionInfo
|
| /external/webrtc/webrtc/base/ |
| testclient.cc | 23 packets_ = new std::vector<Packet*>();
30 for (unsigned i = 0; i < packets_->size(); i++)
31 delete (*packets_)[i];
32 delete packets_;
70 if (packets_->size() != 0) {
80 if (packets_->size() > 0) {
81 packet = packets_->front();
82 packets_->erase(packets_->begin());
125 packets_->push_back(new Packet(remote_addr, buf, size))
[all...] |
| testclient.h | 90 std::vector<Packet*>* packets_; member in class:rtc::TestClient
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_format_h264.cc | 166 assert(packets_.empty());
202 packets_.push(Packet(offset,
224 packets_.push(Packet(fragment_offset,
248 packets_.back().last_fragment = true;
256 if (packets_.empty()) {
262 Packet packet = packets_.front();
268 packets_.pop();
277 *last_packet = packets_.empty();
283 Packet packet = packets_.front();
298 packets_.pop()
[all...] |
| rtp_format_h264.h | 86 PacketQueue packets_; member in class:webrtc::RtpPacketizerH264
|
| rtp_format_vp9.h | 92 PacketInfoQueue packets_; member in class:webrtc::RtpPacketizerVp9
|
| rtp_format_vp9.cc | 537 while (!packets_.empty())
538 packets_.pop();
542 rem_bytes == packet_bytes, &packets_);
551 if (packets_.empty()) {
554 PacketInfo packet_info = packets_.front();
555 packets_.pop();
561 packets_.empty() && (hdr_.spatial_idx == kNoSpatialIdx ||
|
| rtp_format_vp8.h | 212 InfoQueue packets_; member in class:webrtc::RtpPacketizerVp8
|
| rtp_format_vp8.cc | 214 if (packets_.empty()) {
217 InfoStruct packet_info = packets_.front();
218 packets_.pop();
226 *last_packet = packets_.empty();
458 packets_.push(packet_info);
|
| /external/webrtc/webrtc/modules/pacing/ |
| paced_sender.cc | 262 packets_(new paced_sender::PacketQueue(clock)),
312 packets_->Push(paced_sender::Packet(priority, ssrc, sequence_number,
320 return static_cast<int64_t>(packets_->SizeInBytes() * 8 / max_bitrate_kbps_);
325 return packets_->SizeInPackets();
331 int64_t oldest_packet = packets_->OldestEnqueueTimeMs();
340 packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
341 return packets_->AverageQueueTimeMs();
364 size_t queue_size_bytes = packets_->SizeInBytes();
369 packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
371 1, kMaxQueueLengthMs - packets_->AverageQueueTimeMs())
[all...] |
| paced_sender.h | 160 rtc::scoped_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
|
| /external/webrtc/talk/app/webrtc/ |
| datachannel.cc | 90 return packets_.empty();
94 return packets_.front();
98 if (packets_.empty()) {
102 byte_count_ -= packets_.front()->size();
103 packets_.pop_front();
108 packets_.push_back(packet);
112 while (!packets_.empty()) {
113 delete packets_.front();
114 packets_.pop_front();
124 other->packets_.swap(packets_)
[all...] |
| datachannel.h | 230 std::deque<DataBuffer*> packets_; member in class:webrtc::DataChannel::PacketQueue
|
| /external/webrtc/webrtc/test/ |
| rtp_file_reader.cc | 260 packets_(),
297 } else if (result == kResultSuccess && packets_.size() == 1) {
299 PacketIterator it = packets_.begin();
311 printf("Total RTP/RTCP packets: %" PRIuS "\n", packets_.size());
317 uint8_t pt = packets_[packet_indices[0]].rtp_header.payloadType;
338 next_packet_it_ = packets_.begin();
356 if (next_packet_it_ == packets_.end()) {
459 packets_.push_back(marker);
469 static_cast<uint32_t>(packets_.size()));
470 packets_.push_back(marker)
640 std::vector<RtpPacketMarker> packets_; member in class:webrtc::test::PcapReader
[all...] |
| /external/webrtc/webrtc/p2p/base/ |
| dtlstransportchannel.cc | 47 packets_(kMaxPendingPackets, kMaxDtlsPacketLen) {
59 if (!packets_.ReadFront(buffer, buffer_len, read)) {
83 bool ret = packets_.WriteBack(data, size, NULL);
|
| dtlstransportchannel.h | 50 rtc::BufferQueue packets_; member in class:cricket::StreamInterfaceChannel
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| bwe_test.h | 143 Packets packets_; member in class:webrtc::testing::bwe::BweTest
|
| bwe_test.cc | 110 for (Packet* packet : packets_)
181 link->Run(simulation_interval_ms_, time_now_ms_, &packets_);
[all...] |
