| /external/webrtc/webrtc/call/ |
| rtc_event_log2rtp_dump.cc | 164 event.rtcp_packet().has_type() &&
165 event.rtcp_packet().has_packet_data() &&
166 event.rtcp_packet().packet_data().size() > 0) {
167 const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); local
168 if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
170 if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
172 if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA)
178 rtcp_packet.packet_data().data() + 4));
184 packet.length = rtcp_packet.packet_data().size()
[all...] |
| rtc_event_log_unittest.cc | 25 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
254 const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); local
255 ASSERT_TRUE(rtcp_packet.has_incoming());
256 EXPECT_EQ(incoming, rtcp_packet.incoming());
257 ASSERT_TRUE(rtcp_packet.has_type());
258 EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
259 ASSERT_TRUE(rtcp_packet.has_packet_data());
260 ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
262 EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]))
[all...] |
| /external/webrtc/webrtc/video/ |
| 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);
|
| vie_receiver.cc | 229 int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet,
231 return InsertRTCPPacket(static_cast<const uint8_t*>(rtcp_packet),
410 int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
419 rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
422 int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
|
| vie_channel.h | 209 int32_t ReceivedRTCPPacket(const void* rtcp_packet,
|
| vie_channel.cc | [all...] |
| /external/webrtc/talk/media/base/ |
| rtpdump_unittest.cc | 67 RtpDumpPacket rtcp_packet(rtcp_buf.Data(), rtcp_buf.Length(), 0, true);
69 EXPECT_TRUE(rtcp_packet.is_rtcp());
70 EXPECT_FALSE(rtcp_packet.IsValidRtpPacket());
71 EXPECT_TRUE(rtcp_packet.IsValidRtcpPacket());
72 EXPECT_TRUE(rtcp_packet.GetRtcpType(&rtcp_type));
|
| testutils.cc | 187 RawRtcpPacket rtcp_packet; local
188 result &= rtcp_packet.ReadFromByteBuffer(&buf);
189 result &= rtcp_packet.EqualsTo(kTestRawRtcpPackets[index]);
|
| /external/webrtc/talk/session/media/ |
| srtpfilter_unittest.cc | 97 char rtcp_packet[sizeof(kRtcpReport) + 4 + 10]; local
105 memcpy(rtcp_packet, kRtcpReport, rtcp_len);
123 EXPECT_TRUE(f1_.ProtectRtcp(rtcp_packet, rtcp_len,
124 sizeof(rtcp_packet), &out_len));
126 EXPECT_NE(0, memcmp(rtcp_packet, kRtcpReport, rtcp_len));
127 EXPECT_TRUE(f2_.UnprotectRtcp(rtcp_packet, out_len, &out_len));
129 EXPECT_EQ(0, memcmp(rtcp_packet, kRtcpReport, rtcp_len));
131 EXPECT_TRUE(f2_.ProtectRtcp(rtcp_packet, rtcp_len,
132 sizeof(rtcp_packet), &out_len));
134 EXPECT_NE(0, memcmp(rtcp_packet, kRtcpReport, rtcp_len))
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_rtcp_impl.cc | 223 const uint8_t* rtcp_packet,
226 RTCPUtility::RTCPParserV2 rtcp_parser(rtcp_packet, length, true);
|
