/external/webrtc/webrtc/modules/video_coding/ |
generic_encoder.cc | 25 // Map information from info into rtp. If no relevant information is found 26 // in info, rtp is set to NULL. 27 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) { 31 rtp->codec = kRtpVideoVp8; 32 rtp->codecHeader.VP8.InitRTPVideoHeaderVP8(); 33 rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId; 34 rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference; 35 rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx; 36 rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync; 37 rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx [all...] |
/external/webrtc/webrtc/video/ |
vie_remb_unittest.cc | 47 MockRtpRtcp rtp; local 48 vie_remb_->AddReceiveChannel(&rtp); 49 vie_remb_->AddRembSender(&rtp); 58 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs)) 63 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, ssrcs)) 67 vie_remb_->RemoveReceiveChannel(&rtp); 68 vie_remb_->RemoveRembSender(&rtp); 72 MockRtpRtcp rtp; local 73 vie_remb_->AddReceiveChannel(&rtp); 74 vie_remb_->AddRembSender(&rtp); 200 MockRtpRtcp rtp; local 231 MockRtpRtcp rtp; local [all...] |
send_statistics_proxy_unittest.cc | 40 config.rtp.ssrcs.push_back(17); 41 config.rtp.ssrcs.push_back(42); 42 config.rtp.rtx.ssrcs.push_back(18); 43 config.rtp.rtx.ssrcs.push_back(43); 101 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); 102 it != config_.rtp.ssrcs.end(); 115 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); 116 it != config_.rtp.rtx.ssrcs.end(); 159 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); 160 it != config_.rtp.ssrcs.end() [all...] |
video_receive_stream.cc | 56 ss << ", rtp: " << rtp.ToString(); 71 std::string VideoReceiveStream::Config::Rtp::ToString() const { 157 config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions); 174 vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, 176 RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff) 179 vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode); 181 RTC_DCHECK(config_.rtp.remote_ssrc != 0); 183 RTC_DCHECK(config_.rtp.local_ssrc != 0) [all...] |
video_send_stream.cc | 49 std::string VideoSendStream::Config::Rtp::Rtx::ToString() 65 std::string VideoSendStream::Config::Rtp::ToString() const { 94 ss << ", rtp: " << rtp.ToString(); 133 RTC_DCHECK(!config_.rtp.ssrcs.empty()); 137 for (const RtpExtension& extension : config.rtp.extensions) { 145 const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs; 173 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { 174 const std::string& extension = config_.rtp.extensions[i].name; 175 int id = config_.rtp.extensions[i].id [all...] |
payload_router_unittest.cc | 37 MockRtpRtcp rtp; local 38 std::list<RtpRtcp*> modules(1, &rtp); 46 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, 53 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, 60 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, 67 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL, 75 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
|
end_to_end_tests.cc | 65 ADD_FAILURE() << "Unexpected RTP sent."; 306 send_config->rtp.nack.rtp_history_ms = 307 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 439 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 440 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 538 // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms; 539 // send_config->rtp.nack.rtp_history_ms = rtp_history_ms; 540 send_config->rtp.fec.red_payload_type = kRedPayloadType; 541 send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; 543 (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType [all...] |
replay.cc | 89 DEFINE_int32(abs_send_time_id, -1, "RTP extension ID for abs-send-time"); 98 "RTP extension ID for transmission-offset"); 221 receive_config.rtp.remote_ssrc = flags::Ssrc(); 222 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; 223 receive_config.rtp.fec.ulpfec_payload_type = flags::FecPayloadType(); 224 receive_config.rtp.fec.red_payload_type = flags::RedPayloadType(); 225 receive_config.rtp.nack.rtp_history_ms = 1000; 227 receive_config.rtp.extensions.push_back( 231 receive_config.rtp.extensions.push_back(
|
/external/libvorbis/doc/ |
a2-encapsulation-rtp.tex | 4 \section{Vorbis encapsulation in RTP} \label{vorbis:over:rtp} 6 % TODO: Include draft-rtp.xml somehow? 8 Please consult RFC 5215 \textit{``RTP Payload Format for Vorbis Encoded 9 Audio''} for description of how to embed Vorbis audio in an RTP stream.
|
/external/webrtc/webrtc/call/ |
bitrate_estimator_tests.cc | 123 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]); 133 receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0]; 134 receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc; 135 receive_config_.rtp.remb = true; 136 receive_config_.rtp.extensions.push_back( 138 receive_config_.rtp.extensions.push_back( 172 test_->video_send_config_.rtp.ssrcs[0]++; 186 receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0] [all...] |
rampup_tests.cc | 120 send_config->rtp.extensions.clear(); 127 send_config->rtp.extensions.push_back( 132 send_config->rtp.extensions.push_back(RtpExtension( 137 send_config->rtp.extensions.push_back(RtpExtension( 141 send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs; 142 send_config->rtp.ssrcs = video_ssrcs_; 144 send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType; 145 send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_; 148 send_config->rtp.fec.ulpfec_payload_type = 150 send_config->rtp.fec.red_payload_type = test::CallTest::kRedPayloadType [all...] |
call_unittest.cc | 47 config.rtp.ssrc = 42; 57 config.rtp.remote_ssrc = 42; 71 config.rtp.ssrc = ssrc; 94 config.rtp.remote_ssrc = ssrc;
|
rtc_event_log_unittest.cc | 84 << (event.has_rtp_packet() ? "" : "no ") << "RTP packet"; 131 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc()); 133 EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc()); 136 if (config.rtp.rtcp_mode == RtcpMode::kCompound) 143 EXPECT_EQ(config.rtp.remb, receiver_config.remb()); 145 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()), 150 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type())); 152 const VideoReceiveStream::Config::Rtp::Rtx& rtx = 153 config.rtp.rtx.at(rtx_map.payload_type()); 160 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()) [all...] |
/frameworks/av/media/libstagefright/wifi-display/rtp/ |
RTPSender.cpp | 219 uint8_t *rtp = udpPacket->data(); local 220 rtp[0] = 0x80; 221 rtp[1] = packetType; 223 rtp[2] = (mRTPSeqNo >> 8) & 0xff; 224 rtp[3] = mRTPSeqNo & 0xff; 229 rtp[4] = rtpTime >> 24; 230 rtp[5] = (rtpTime >> 16) & 0xff; 231 rtp[6] = (rtpTime >> 8) & 0xff; 232 rtp[7] = rtpTime & 0xff; 234 rtp[8] = kSourceID >> 24 262 uint8_t *rtp = udpPacket->data(); local [all...] |
/cts/tests/tests/net/src/android/net/rtp/cts/ |
AudioStreamTest.java | 16 package android.net.rtp.cts; 18 import android.net.rtp.AudioCodec; 19 import android.net.rtp.AudioStream;
|
AudioGroupTest.java | 16 package android.net.rtp.cts; 20 import android.net.rtp.AudioCodec; 21 import android.net.rtp.AudioGroup; 22 import android.net.rtp.AudioStream; 23 import android.net.rtp.RtpStream;
|
AudioCodecTest.java | 16 package android.net.rtp.cts; 18 import android.net.rtp.AudioCodec;
|
/external/srtp/test/ |
rtpw.c | 4 * rtp word sender/receiver 9 * This app is a simple RTP application intended only for testing 12 * each USEC_RATE microseconds. Secure RTP protections can be 79 #include "rtp.h" 119 * program_type distinguishes the [s]rtp sender and receiver cases 314 crypto_policy_set_rtp_default(&policy.rtp); 318 crypto_policy_set_aes_cm_128_null_auth(&policy.rtp); 322 crypto_policy_set_null_cipher_hmac_sha1_80(&policy.rtp); 335 policy.rtp.sec_serv = sec_servs; 370 * application is now a vanilla-flavored RTP application [all...] |
/external/webrtc/webrtc/audio/ |
audio_receive_stream.cc | 37 if (!config.rtp.transport_cc) { 40 for (const auto& extension : config.rtp.extensions) { 49 std::string AudioReceiveStream::Config::Rtp::ToString() const { 67 ss << "{rtp: " << rtp.ToString(); 98 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); 99 for (const auto& extension : config.rtp.extensions) { 115 RTC_NOTREACHED() << "Unsupported RTP extension."; 138 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); 175 // bandwidth estimation. RTP timestamps has different rates for audio an [all...] |
audio_send_stream_unittest.cc | 97 stream_config_.rtp.ssrc = kSsrc; 98 stream_config_.rtp.c_name = kCName; 99 stream_config_.rtp.extensions.push_back( 101 stream_config_.rtp.extensions.push_back( 103 stream_config_.rtp.extensions.push_back(RtpExtension( 170 config.rtp.ssrc = kSsrc; 171 config.rtp.extensions.push_back( 173 config.rtp.c_name = kCName; 178 "{rtp: {ssrc: 1234, extensions: [{name: " 179 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], [all...] |
audio_receive_stream_unittest.cc | 104 stream_config_.rtp.local_ssrc = kLocalSsrc; 105 stream_config_.rtp.remote_ssrc = kRemoteSsrc; 106 stream_config_.rtp.extensions.push_back( 108 stream_config_.rtp.extensions.push_back( 127 RemoveStream(stream_config_.rtp.remote_ssrc)); 206 config.rtp.remote_ssrc = kRemoteSsrc; 207 config.rtp.local_ssrc = kLocalSsrc; 208 config.rtp.extensions.push_back( 213 "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: " 214 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, [all...] |
/external/webrtc/talk/media/webrtc/ |
webrtcvoe.h | 102 webrtc::VoERTP_RTCP* rtp, 110 rtp_(rtp), 120 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } function in class:cricket::VoEWrapper
|
/external/webrtc/webrtc/ |
audio_send_stream.h | 60 // Receive-stream specific RTP settings. 61 struct Rtp { 67 // RTP header extensions used for the sent stream. 72 } rtp; member in struct:webrtc::AudioSendStream::Config
|
/frameworks/opt/net/voip/src/java/android/net/rtp/ |
AudioStream.java | 17 package android.net.rtp; 24 * Real-time Transport Protocol (RTP). Two different classes are developed in 130 * Returns the RTP payload type for dual-tone multi-frequency (DTMF) digits, 140 * Sets the RTP payload type for dual-tone multi-frequency (DTMF) digits. 143 * RTP payload type for DTMF is assigned dynamically, so it must be in the 148 * @param type The RTP payload type to be used or {@code -1} to disable it.
|
/external/webrtc/webrtc/test/ |
call_test.cc | 190 video_send_config_.rtp.extensions.push_back( 194 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); 195 video_send_config_.rtp.extensions.push_back(RtpExtension( 202 audio_send_config_.rtp.ssrc = kAudioSendSsrc; 210 RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty()); 212 video_config.rtp.remb = true; 213 video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc; 214 for (const RtpExtension& extension : video_send_config_.rtp.extensions) 215 video_config.rtp.extensions.push_back(extension); 216 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) [all...] |