/external/webrtc/webrtc/modules/rtp_rtcp/ |
D | rtp_rtcp.gypi | 42 'source/rtcp_packet.cc', 43 'source/rtcp_packet.h', 44 'source/rtcp_packet/app.cc', 45 'source/rtcp_packet/app.h', 46 'source/rtcp_packet/bye.cc', 47 'source/rtcp_packet/bye.h', 48 'source/rtcp_packet/compound_packet.cc', 49 'source/rtcp_packet/compound_packet.h', 50 'source/rtcp_packet/dlrr.cc', 51 'source/rtcp_packet/dlrr.h', [all …]
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D | BUILD.gn | 47 "source/rtcp_packet.cc", 48 "source/rtcp_packet.h", 49 "source/rtcp_packet/app.cc", 50 "source/rtcp_packet/app.h", 51 "source/rtcp_packet/bye.cc", 52 "source/rtcp_packet/bye.h", 53 "source/rtcp_packet/compound_packet.cc", 54 "source/rtcp_packet/compound_packet.h", 55 "source/rtcp_packet/dlrr.cc", 56 "source/rtcp_packet/dlrr.h", [all …]
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/external/webrtc/webrtc/call/ |
D | rtc_event_log2rtp_dump.cc | 164 event.rtcp_packet().has_type() && in main() 165 event.rtcp_packet().has_packet_data() && in main() 166 event.rtcp_packet().packet_data().size() > 0) { in main() 167 const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); in main() local 168 if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO) in main() 170 if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO) in main() 172 if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA) in main() 178 rtcp_packet.packet_data().data() + 4)); in main() 184 packet.length = rtcp_packet.packet_data().size(); in main() 195 memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length); in main()
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D | rtc_event_log_unittest.cc | 254 const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); in VerifyRtcpEvent() local 255 ASSERT_TRUE(rtcp_packet.has_incoming()); in VerifyRtcpEvent() 256 EXPECT_EQ(incoming, rtcp_packet.incoming()); in VerifyRtcpEvent() 257 ASSERT_TRUE(rtcp_packet.has_type()); in VerifyRtcpEvent() 258 EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type())); in VerifyRtcpEvent() 259 ASSERT_TRUE(rtcp_packet.has_packet_data()); in VerifyRtcpEvent() 260 ASSERT_EQ(total_size, rtcp_packet.packet_data().size()); in VerifyRtcpEvent() 262 EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i])); in VerifyRtcpEvent()
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D | rtc_event_log.proto | 52 optional RtcpPacket rtcp_packet = 4; field
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/external/webrtc/webrtc/modules/ |
D | modules.gyp | 308 'rtp_rtcp/source/rtcp_packet/app_unittest.cc', 309 'rtp_rtcp/source/rtcp_packet/bye_unittest.cc', 310 'rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc', 311 'rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc', 312 'rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc', 313 'rtp_rtcp/source/rtcp_packet/nack_unittest.cc', 314 'rtp_rtcp/source/rtcp_packet/pli_unittest.cc', 315 'rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc', 316 'rtp_rtcp/source/rtcp_packet/report_block_unittest.cc', 317 'rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc', [all …]
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/external/webrtc/talk/media/base/ |
D | rtpdump_unittest.cc | 67 RtpDumpPacket rtcp_packet(rtcp_buf.Data(), rtcp_buf.Length(), 0, true); in TEST() local 69 EXPECT_TRUE(rtcp_packet.is_rtcp()); in TEST() 70 EXPECT_FALSE(rtcp_packet.IsValidRtpPacket()); in TEST() 71 EXPECT_TRUE(rtcp_packet.IsValidRtcpPacket()); in TEST() 72 EXPECT_TRUE(rtcp_packet.GetRtcpType(&rtcp_type)); in TEST()
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D | testutils.cc | 187 RawRtcpPacket rtcp_packet; in VerifyTestPacketsFromStream() local 188 result &= rtcp_packet.ReadFromByteBuffer(&buf); in VerifyTestPacketsFromStream() 189 result &= rtcp_packet.EqualsTo(kTestRawRtcpPackets[index]); in VerifyTestPacketsFromStream()
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/external/webrtc/webrtc/video/ |
D | vie_receiver.cc | 229 int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet, in ReceivedRTCPPacket() argument 231 return InsertRTCPPacket(static_cast<const uint8_t*>(rtcp_packet), in ReceivedRTCPPacket() 410 int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet, in InsertRTCPPacket() argument 419 rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); in InsertRTCPPacket() 422 int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); in InsertRTCPPacket()
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D | vie_receiver.h | 79 int ReceivedRTCPPacket(const void* rtcp_packet, size_t rtcp_packet_length); 102 int InsertRTCPPacket(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
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D | vie_channel.h | 209 int32_t ReceivedRTCPPacket(const void* rtcp_packet,
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D | vie_channel.cc | 942 int32_t ViEChannel::ReceivedRTCPPacket(const void* rtcp_packet, in ReceivedRTCPPacket() argument 944 return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length); in ReceivedRTCPPacket()
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/external/webrtc/talk/session/media/ |
D | srtpfilter_unittest.cc | 97 char rtcp_packet[sizeof(kRtcpReport) + 4 + 10]; in TestProtectUnprotect() local 105 memcpy(rtcp_packet, kRtcpReport, rtcp_len); in TestProtectUnprotect() 123 EXPECT_TRUE(f1_.ProtectRtcp(rtcp_packet, rtcp_len, in TestProtectUnprotect() 124 sizeof(rtcp_packet), &out_len)); in TestProtectUnprotect() 126 EXPECT_NE(0, memcmp(rtcp_packet, kRtcpReport, rtcp_len)); in TestProtectUnprotect() 127 EXPECT_TRUE(f2_.UnprotectRtcp(rtcp_packet, out_len, &out_len)); in TestProtectUnprotect() 129 EXPECT_EQ(0, memcmp(rtcp_packet, kRtcpReport, rtcp_len)); in TestProtectUnprotect() 131 EXPECT_TRUE(f2_.ProtectRtcp(rtcp_packet, rtcp_len, in TestProtectUnprotect() 132 sizeof(rtcp_packet), &out_len)); in TestProtectUnprotect() 134 EXPECT_NE(0, memcmp(rtcp_packet, kRtcpReport, rtcp_len)); in TestProtectUnprotect() [all …]
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtp_rtcp_impl.cc | 223 const uint8_t* rtcp_packet, in IncomingRtcpPacket() argument 226 RTCPUtility::RTCPParserV2 rtcp_parser(rtcp_packet, length, true); in IncomingRtcpPacket()
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