/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | producer_fec_unittest.cc | 24 RedPacket* packet, 26 EXPECT_GT(packet->length(), static_cast<int>(kRtpHeaderSize)); 27 EXPECT_TRUE(packet->data() != NULL); 28 uint8_t* data = packet->data(); 68 producer_->SetFecParameters(¶ms, 0); // Expecting one FEC packet. 80 RedPacket* packet = producer_->GetFecPacket(kRedPayloadType, local 85 ASSERT_TRUE(packet != NULL); 87 kRedPayloadType, kFecPayloadType, packet, false); 92 delete packet; 108 producer_->SetFecParameters(¶ms, 0); // Expecting one FEC packet 20 VerifyHeader(uint16_t seq_num, uint32_t timestamp, int red_pltype, int fec_pltype, RedPacket* packet, bool marker_bit) argument 123 RedPacket* packet = producer_->GetFecPacket(kRedPayloadType, local 140 RtpPacket* packet = generator_->NextPacket(0, 10); local [all...] |
H A D | rtcp_format_remb_unittest.cc | 37 const void *packet, 39 RTCPUtility::RTCPParserV2 rtcpParser((uint8_t*)packet, 36 SendRTCPPacket(int , const void *packet, int packetLength) argument
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H A D | rtcp_packet.cc | 110 void CreateHeader(uint8_t count_or_format, // Depends on packet type. 136 // | sender's packet count | 160 // | SSRC of packet sender | 273 // Bye packet (BYE) (RFC 3550). 299 // Application-Defined packet (APP) (RFC 3550). 327 // Common packet format: 334 // | SSRC of packet sender | 539 // | SSRC of packet sender | 725 void RtcpPacket::Append(RtcpPacket* packet) { argument 726 assert(packet); 732 uint8_t packet[IP_PACKET_SIZE]; local 737 Build(uint8_t* packet, size_t* length, size_t max_length) const argument 744 CreateAndAddAppended(uint8_t* packet, size_t* length, size_t max_length) const argument 754 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 757 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 778 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 799 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 815 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 854 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 870 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 878 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 886 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 895 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 929 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 966 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 975 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 992 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 1014 Create(uint8_t* packet, size_t* length, size_t max_length) const argument 1023 Create(uint8_t* packet, size_t* length, size_t max_length) const argument [all...] |
H A D | rtcp_packet.h | 51 // uint8_t packet[kPacketSize]; // with sequence number 123. 52 // fir.Build(packet, &length, kPacketSize); 54 // RawPacket packet = fir.Build(); // Returns a RawPacket holding 55 // // the built rtcp packet. 57 // rr.Append(&fir) // Builds a compound RTCP packet with 58 // RawPacket packet = rr.Build(); // a receiver report, report block 65 void Append(RtcpPacket* packet); 69 void Build(uint8_t* packet, size_t* length, size_t max_length) const; 75 uint8_t* packet, size_t* length, size_t max_length) const = 0; 81 uint8_t* packet, size_ 1066 RawPacket(const uint8_t* packet, size_t length) argument [all...] |
H A D | rtcp_packet_unittest.cc | 51 RawPacket packet = rr.Build(); local 53 parser.Parse(packet.buffer(), packet.buffer_length()); 73 RawPacket packet = rr.Build(); local 75 parser.Parse(packet.buffer(), packet.buffer_length()); 99 RawPacket packet = rr.Build(); local 101 parser.Parse(packet.buffer(), packet.buffer_length()); 118 RawPacket packet local 140 RawPacket packet = sr.Build(); local 160 RawPacket packet = sr.Build(); local 173 RawPacket packet = ij.Build(); local 184 RawPacket packet = ij.Build(); local 197 RawPacket packet = ij.Build(); local 214 RawPacket packet = app.Build(); local 236 RawPacket packet = app.Build(); local 252 RawPacket packet = sdes.Build(); local 270 RawPacket packet = sdes.Build(); local 283 RawPacket packet = sdes.Build(); local 297 RawPacket packet = pli.Build(); local 316 RawPacket packet = sli.Build(); local 335 RawPacket packet = nack.Build(); local 356 RawPacket packet = nack.Build(); local 378 RawPacket packet = rpsi.Build(); local 393 RawPacket packet = rpsi.Build(); local 407 RawPacket packet = rpsi.Build(); local 421 RawPacket packet = rpsi.Build(); local 436 RawPacket packet = rpsi.Build(); local 449 RawPacket packet = fir.Build(); local 467 RawPacket packet = rr.Build(); local 482 RawPacket packet = empty.Build(); local 502 RawPacket packet = sr.Build(); local 516 RawPacket packet = bye.Build(); local 531 RawPacket packet = bye.Build(); local 552 uint8_t packet[kRrLength + kReportBlockLength + kFirLength]; local 573 uint8_t packet[kRrLength + kReportBlockLength - 1]; local 592 uint8_t packet[kRrLength + kReportBlockLength - 1]; local 609 RawPacket packet = remb.Build(); local 629 RawPacket packet = tmmbr.Build(); local 643 RawPacket packet = tmmbn.Build(); local 656 RawPacket packet = tmmbn.Build(); local 673 RawPacket packet = tmmbn.Build(); local 691 RawPacket packet = xr.Build(); local 706 RawPacket packet = xr.Build(); local 728 RawPacket packet = xr.Build(); local 745 RawPacket packet = xr.Build(); local 765 RawPacket packet = xr.Build(); local 790 RawPacket packet = xr.Build(); local 833 RawPacket packet = xr.Build(); local 873 RawPacket packet = xr.Build(); local 894 RawPacket packet = xr.Build(); local [all...] |
H A D | rtcp_receiver.cc | 245 *RTCPArrivalTimeFrac = _lastReceivedSRNTPfrac; // local NTP time when we received a RTCP packet with a send block 326 // next top level packet. 409 // The synchronization source identifier for the originator of this SR packet 412 // The source of the packet sender, same as of SR? or is this a CE? 483 // This will be called once per report block in the RTCP packet. 485 // Each packet has max 31 RR blocks. 729 // time since last received rtcp packet 737 // no rtcp packet for the last five regular intervals, reset limitations 898 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local 900 rtcpPacketInformation.xr_originator_ssrc = packet 908 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local 927 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local [all...] |
H A D | rtcp_receiver_unittest.cc | 47 // Injects an RTCP packet into the receiver. 48 virtual int SendRTCPPacket(int /* ch */, const void *packet, int packet_len) { argument 94 // Injects an RTCP packet into the receiver. 96 int InjectRtcpPacket(const uint8_t* packet, argument 98 RTCPUtility::RTCPParserV2 rtcpParser(packet, 158 // expected peer, but will not flag that he's gotten a packet. 509 // The parser should note the DLRR report block item, but not flag the packet 529 // The parser should note the DLRR report block item, but not flag the packet 555 // The parser should note the DLRR report block item, but not flag the packet 796 // 5 seconds between each packet [all...] |
H A D | rtcp_sender_unittest.cc | 234 virtual int SendRTCPPacket(int /*ch*/, const void *packet, int packet_len) { argument 235 RTCPUtility::RTCPParserV2 rtcpParser((uint8_t*)packet, 316 // Helper function: Incoming RTCP has a specific packet type. 371 // Make sure RTP packet has been received. 387 // Transmission time offset packet should be received. 398 // Transmission time offset packet should not be received. 446 // No packet sent. 480 // We now expect the packet to show up in the rtcp_packet_info_ of 503 // We now expect the packet to show up in the rtcp_packet_info_ of
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H A D | rtp_fec_unittest.cc | 33 T* packet = NULL; local 35 packet = my_list->front(); 36 delete packet; 55 // Media packet "i" is lost if media_loss_mask_[i] = 1, 59 // FEC packet "i" is lost if fec_loss_mask_[i] = 1, 63 // Construct the media packet list, up to |num_media_packets| packets. 64 // Returns the next sequence number after the last media packet. 65 // (this will be the sequence of the first FEC packet) 69 // Construct the received packet list: a subset of the media and FEC packets. 72 // Add packet fro 849 ForwardErrorCorrection::Packet* packet; local [all...] |
H A D | rtp_format_h264.cc | 162 // Aggregate fragments into one packet (STAP-A). 187 // If we are going to try to aggregate more fragments into this packet 189 // NALU of this packet. 208 Packet packet = packets_.front(); local 210 if (packet.first_fragment && packet.last_fragment) { 211 // Single NAL unit packet. 212 *bytes_to_send = packet.size; 213 memcpy(buffer, &payload_data_[packet.offset], packet 229 Packet packet = packets_.front(); local 254 Packet packet = packets_.front(); local [all...] |
H A D | rtp_format_h264_unittest.cc | 53 const uint8_t* packet, 59 EXPECT_EQ(kFuIndicator, packet[0]) << "FUA index: " << fua_index; 68 EXPECT_EQ(fu_header, packet[1]) << "FUA index: " << fua_index; 74 ::testing::ElementsAreArray(&packet[2], expected_sizes[fua_index])) 95 scoped_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]); 100 ASSERT_TRUE(packetizer->NextPacket(packet.get(), &length, &last)); 101 VerifyFua(i, frame.get(), offset, packet.get(), length, expected_sizes); 106 EXPECT_FALSE(packetizer->NextPacket(packet.get(), &length, &last)); 126 const uint8_t* packet, 140 ::testing::ElementsAreArray(&packet[expected_payload_offse 50 VerifyFua(size_t fua_index, const uint8_t* expected_payload, int offset, const uint8_t* packet, size_t length, const std::vector<size_t>& expected_sizes) argument 198 uint8_t packet[kMaxPayloadSize] = {0}; local 235 uint8_t packet[kMaxPayloadSize] = {0}; local 270 uint8_t packet[kMaxPayloadSize] = {0}; local 322 uint8_t packet[kMaxPayloadSize] = {0}; local 409 uint8_t packet[2] = {0x05, 0xFF}; // F=0, NRI=0, Type=5. local 422 uint8_t packet[16] = {kStapA, // F=0, NRI=0, Type=24. local 438 uint8_t packet[16] = {kStapA, // F=0, NRI=0, Type=24. local [all...] |
H A D | rtp_format_vp8_unittest.cc | 258 // Frag start only true for first packet in equal size mode. 284 // EqualSize mode => First packet full; other not. 287 // Frag start only true for first packet in equal size mode. 317 // Expect one single packet of payload_size() + 4 bytes header. 346 // Expect one single packet of payload_size() + 3 bytes header. 376 // Expect one single packet of payload_size() + 3 bytes header. 410 uint8_t packet[4] = {0}; local 411 packet[0] = 0x14; // Binary 0001 0100; S = 1, PartID = 4. 412 packet[1] = 0x01; // P frame. 417 ExpectPacket(packet 430 uint8_t packet[10] = {0}; local 463 uint8_t packet[13] = {0}; local 481 uint8_t packet[10] = {0}; local 500 uint8_t packet[10] = {0}; local 518 uint8_t packet[10] = {0}; local 537 uint8_t packet[4] = {0}; local 552 uint8_t packet[20] = {0}; local [all...] |
H A D | rtp_header_parser.cc | 24 virtual bool Parse(const uint8_t* packet, 45 bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) { argument 46 RtpUtility::RtpHeaderParser rtp_parser(packet, length); 50 bool RtpHeaderParserImpl::Parse(const uint8_t* packet, argument 53 RtpUtility::RtpHeaderParser rtp_parser(packet, length);
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H A D | rtp_packet_history.cc | 48 LOG(LS_WARNING) << "Purging packet history in order to re-set status."; 114 int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet, argument 128 assert(packet); 134 LOG(LS_WARNING) << "Failed to store RTP packet with length: " 139 const uint16_t seq_num = (packet[2] << 8) + packet[3]; 141 // Store packet 144 std::copy(packet, packet + packet_length, it->begin()); 183 uint8_t* packet, 180 GetPacketAndSetSendTime(uint16_t sequence_number, uint32_t min_elapsed_time_ms, bool retransmit, uint8_t* packet, uint16_t* packet_length, int64_t* stored_time_ms) argument 224 GetPacket(int index, uint8_t* packet, uint16_t* packet_length, int64_t* stored_time_ms) const argument 238 GetBestFittingPacket(uint8_t* packet, uint16_t* packet_length, int64_t* stored_time_ms) argument [all...] |
H A D | rtp_payload_registry.cc | 244 const uint8_t* packet, 251 const uint8_t* rtx_header = packet + header.headerLength; 254 // Copy the packet into the restored packet, except for the RTX header. 255 memcpy(*restored_packet, packet, header.headerLength); 257 packet + header.headerLength + kRtxHeaderSize, 276 LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet."; 243 RestoreOriginalPacket(uint8_t** restored_packet, const uint8_t* packet, int* packet_length, uint32_t original_ssrc, const RTPHeader& header) const argument
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H A D | rtp_rtcp_impl_unittest.cc | 195 rtcp::RawPacket packet = nack.Build(); local 196 EXPECT_EQ(0, module->impl_->IncomingRtcpPacket(packet.buffer(), 197 packet.buffer_length()));
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H A D | rtp_sender.cc | 412 // Drop this packet if we're not sending media packets. 485 int RTPSender::BuildPaddingPacket(uint8_t* packet, int header_length, argument 491 packet[0] |= 0x20; // Set padding bit. 493 reinterpret_cast<int32_t *>(&(packet[header_length])); 499 // Set number of padding bytes in the last byte of the packet. 500 packet[header_length + padding_bytes_in_packet - 1] = padding_bytes_in_packet; 537 // Only send padding packets following the last packet of a frame, 549 // Without abs-send-time a media packet must be sent before padding so 634 // We can't send the packet right now. 649 bool RTPSender::SendPacketToNetwork(const uint8_t *packet, uint32_ argument [all...] |
H A D | rtp_sender_unittest.cc | 48 const uint8_t* packet) { 49 return packet + rtp_header.headerLength; 474 // Parse sent packet. 536 // Parse sent packet. 554 // This test sends 1 regular video packet, then 4 padding packets, and then 555 // 1 more regular packet. 607 // packet, since it is tested in another test. 623 // Parse sent packet. 639 // Send a regular video packet again. 659 // Parse sent packet 47 GetPayloadData(const RTPHeader& rtp_header, const uint8_t* packet) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/utility/source/ |
H A D | rtp_dump_impl.cc | 50 // Length of packet, including this header (may be smaller than plen if not 51 // whole packet recorded). 148 int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength) argument 156 if (packet == NULL) 166 // If the packet doesn't contain a valid RTCP header the packet will be 168 bool isRTCP = RTCP(packet); 199 if (!_file.Write(packet, packetLength)) 208 bool RtpDumpImpl::RTCP(const uint8_t* packet) const 210 const uint8_t payloadType = packet[ [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test/ |
H A D | packet_manipulator.cc | 48 uint8_t* packet = NULL; local 53 while ((nbr_bytes_to_read = packet_reader_->NextPacket(&packet)) > 0) { 54 // Check if we're currently in a packet loss burst that is not completed:
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/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test_framework/ |
H A D | packet_loss_test.cc | 115 printf("Target packet loss rate: %.4f\n", _lossProbability); 116 printf("Actual packet loss rate: %.4f\n", (_totalThrown * 1.0f) / (_totalKept + _totalThrown)); 122 printf("No packet losses inflicted\n"); 179 // Only packet loss for delta frames 187 unsigned char *packet = NULL; local 195 while ((size = NextPacket(1500, &packet)) > 0) 199 InsertPacket(&newEncBuf, packet, size); 205 // parts of a packet, and not the whole packet. 207 //int size2 = ByteLoss(size, packet, 1 [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
H A D | decoding_state.cc | 16 #include "webrtc/modules/video_coding/main/source/packet.h" 57 bool VCMDecodingState::IsOldPacket(const VCMPacket* packet) const { 58 assert(packet != NULL); 61 return !IsNewerTimestamp(packet->timestamp, time_stamp_); 102 void VCMDecodingState::UpdateOldPacket(const VCMPacket* packet) { argument 103 assert(packet != NULL); 104 if (packet->timestamp == time_stamp_) { 105 // Late packet belonging to the last decoded frame - make sure we update the 107 sequence_num_ = LatestSequenceNumber(packet->seqNum, sequence_num_);
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H A D | decoding_state_unittest.cc | 18 #include "webrtc/modules/video_coding/main/source/packet.h" 34 VCMPacket packet; local 35 packet.isFirstPacket = true; 36 packet.timestamp = 1; 37 packet.seqNum = 0xffff; 38 packet.frameType = kVideoFrameDelta; 39 packet.codecSpecificHeader.codec = kRtpVideoVp8; 40 packet.codecSpecificHeader.codecHeader.VP8.pictureId = 0x007F; 44 EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data)); 48 packet 166 VCMPacket packet; local 212 VCMPacket packet; local 365 VCMPacket packet; local 400 VCMPacket packet; local 419 VCMPacket packet; local [all...] |
H A D | frame_buffer.cc | 16 #include "webrtc/modules/video_coding/main/source/packet.h" 84 // Insert packet 86 VCMFrameBuffer::InsertPacket(const VCMPacket& packet, argument 90 assert(!(NULL == packet.dataPtr && packet.sizeBytes > 0)); 91 if (packet.dataPtr != NULL) { 92 _payloadType = packet.payloadType; 96 // First packet (empty and/or media) inserted into this frame. 98 _timeStamp = packet.timestamp; 99 // We only take the ntp timestamp of the first packet o [all...] |
H A D | jitter_buffer.cc | 23 #include "webrtc/modules/video_coding/main/source/packet.h" 381 // Will the packet sequence be complete if the next frame is grabbed for 544 VCMFrameBufferEnum VCMJitterBuffer::GetFrame(const VCMPacket& packet, argument 546 // Does this packet belong to an old frame? 547 if (last_decoded_state_.IsOldPacket(&packet)) { 549 if (packet.sizeBytes > 0) { 553 // Update last decoded sequence number if the packet arrived late and 556 last_decoded_state_.UpdateOldPacket(&packet); 569 *frame = incomplete_frames_.FindFrame(packet.timestamp); 572 *frame = decodable_frames_.FindFrame(packet 603 InsertPacket(const VCMPacket& packet, bool* retransmitted) argument [all...] |