/external/webrtc/test/fuzzers/ |
D | flexfec_sender_fuzzer.cc | 51 if (i + kRtpHeaderSize + payload_size >= size) in FuzzOneInput() 54 new uint8_t[kRtpHeaderSize + payload_size]); in FuzzOneInput() 55 memcpy(packet.get(), &data[i], kRtpHeaderSize + payload_size); in FuzzOneInput() 56 i += kRtpHeaderSize + payload_size; in FuzzOneInput() 60 if (!rtp_packet.Parse(packet.get(), kRtpHeaderSize + payload_size)) in FuzzOneInput()
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D | ulpfec_receiver_fuzzer.cc | 44 size_t packet_length = kRtpHeaderSize + fuzz_data.Read<uint8_t>(); in FuzzOneInput()
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D | flexfec_receiver_fuzzer.cc | 48 packet_length = kRtpHeaderSize + data[i++]; in FuzzOneInput()
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/external/webrtc/modules/rtp_rtcp/source/ |
D | fec_test_helper.cc | 124 packet->data.SetSize(length + kRtpHeaderSize); in NextPacket() 127 data[i + kRtpHeaderSize] = offset + i; in NextPacket() 128 packet->data.SetSize(length + kRtpHeaderSize); in NextPacket() 129 packet->header.headerLength = kRtpHeaderSize; in NextPacket() 162 static_cast<size_t>(IP_PACKET_SIZE - kRtpHeaderSize)); in BuildFlexfecPacket() 173 packet_with_rtp_header->data.SetSize(kRtpHeaderSize + packet.data.size()); in BuildFlexfecPacket() 175 memcpy(packet_with_rtp_header->data.data() + kRtpHeaderSize, in BuildFlexfecPacket()
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D | forward_error_correction.cc | 131 if (media_packet->data.size() < kRtpHeaderSize) { in EncodeFec() 223 media_packet->data.size() - kRtpHeaderSize; in GenerateFecPayloads() 245 memcpy(&data[fec_header_size], &media_packet_data[kRtpHeaderSize], in GenerateFecPayloads() 556 std::min(size_t{IP_PACKET_SIZE - kRtpHeaderSize}, in StartPacketRecovery() 564 kRtpHeaderSize); in StartPacketRecovery() 571 kRtpHeaderSize); in StartPacketRecovery() 574 memcpy(recovered_packet->pkt->data.data() + kRtpHeaderSize, in StartPacketRecovery() 590 ByteReader<uint16_t>::ReadBigEndian(&data[2]) + kRtpHeaderSize; in FinishPacketRecovery() 591 if (new_size > size_t{IP_PACKET_SIZE - kRtpHeaderSize}) { in FinishPacketRecovery() 615 src.data.size() - kRtpHeaderSize); in XorHeaders() [all …]
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D | ulpfec_generator_unittest.cc | 47 EXPECT_EQ(static_cast<uint8_t>(fec_payload_type), data[kRtpHeaderSize]); in VerifyHeader() 133 EXPECT_EQ(fec_packets[0]->headers_size(), kRtpHeaderSize); in TEST_F()
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D | flexfec_sender_unittest.cc | 103 EXPECT_EQ(kRtpHeaderSize, fec_packet->headers_size()); in TEST() 143 EXPECT_EQ(kRtpHeaderSize, fec_packet->headers_size()); in TEST() 181 EXPECT_EQ(kRtpHeaderSize, fec_packet->headers_size()); in TEST()
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D | flexfec_receiver.cc | 93 RTC_DCHECK_GE(packet.size(), kRtpHeaderSize); in AddReceivedPacket()
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D | ulpfec_generator.cc | 141 RTC_DCHECK_GE(packet.headers_size(), kRtpHeaderSize); in AddPacketAndGenerateFec()
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D | rtp_sender_audio.cc | 359 new RtpPacketToSend(kNoExtensions, kRtpHeaderSize + kDtmfSize)); in SendTelephoneEventPacket()
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D | rtp_sender.cc | 173 max_media_packet_header_(kRtpHeaderSize), in RTPSender() 174 max_padding_fec_packet_header_(kRtpHeaderSize), in RTPSender()
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D | rtp_fec_unittest.cc | 59 kRtpHeaderSize, // Minimum packet size. in RtpFecTest() 60 IP_PACKET_SIZE - kRtpHeaderSize - kTransportOverhead - in RtpFecTest()
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D | ulpfec_receiver_unittest.cc | 119 packet_generator_.NextPacket(frame_offset + i, kRtpHeaderSize + 10)); in PacketizeFrame()
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D | rtp_sender_video.cc | 212 rtp_sender_->FecOrPaddingPacketMaxRtpHeaderLength() - kRtpHeaderSize; in FecPacketOverhead()
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D | rtp_sender_unittest.cc | 577 EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _)) in TEST_P() 583 EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _)) in TEST_P() 1097 size_t rtp_header_len = kRtpHeaderSize; in TEST_P()
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/external/webrtc/modules/audio_coding/acm2/ |
D | acm_send_test.cc | 142 const size_t kRtpHeaderSize = 12; in CreatePacket() local 143 size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize; in CreatePacket() 163 memcpy(packet_memory + kRtpHeaderSize, &last_payload_vec_[0], in CreatePacket()
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/external/webrtc/call/ |
D | flexfec_receive_stream_unittest.cc | 153 OnRecoveredPacket(_, kRtpHeaderSize + kPayloadLength[1])); in TEST_F()
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/external/webrtc/audio/ |
D | audio_receive_stream_unittest.cc | 196 header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength + in CreateRtpHeaderWithOneByteExtension() 206 BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id, in CreateRtpHeaderWithOneByteExtension()
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/external/webrtc/video/ |
D | video_send_stream.cc | 35 size_t header_size = kRtpHeaderSize; in CalculateMaxHeaderSize()
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/external/webrtc/modules/rtp_rtcp/include/ |
D | rtp_rtcp_defines.h | 48 const uint8_t kRtpHeaderSize = 12; variable
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/external/webrtc/media/engine/ |
D | webrtc_video_engine_unittest.cc | 99 constexpr uint32_t kRtpHeaderSize = 12; variable 1766 NumRtpBytes() - kRtpHeaderSize * NumRtpPackets()); in TEST_F() 1789 EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(), in TEST_F() 1843 EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(), in TEST_F() 1853 EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(), in TEST_F()
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