/external/webrtc/webrtc/audio/ |
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...] |
audio_send_stream.cc | 31 std::string AudioSendStream::Config::Rtp::ToString() const { 49 ss << "{rtp: " << rtp.ToString(); 76 channel_proxy_->SetLocalSSRC(config.rtp.ssrc); 77 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); 79 for (const auto& extension : config.rtp.extensions) { 87 RTC_NOTREACHED() << "Registering unsupported RTP extension."; 128 stats.local_ssrc = config_.rtp.ssrc;
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/external/webrtc/webrtc/video/ |
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(
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send_statistics_proxy.cc | 221 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) == 222 config_.rtp.ssrcs.end() && 223 std::find(config_.rtp.rtx.ssrcs.begin(), 224 config_.rtp.rtx.ssrcs.end(), 225 ssrc) == config_.rtp.rtx.ssrcs.end()) { 254 if (simulcast_idx >= config_.rtp.ssrcs.size()) { 256 << " >= " << config_.rtp.ssrcs.size() << ")."; 259 uint32_t ssrc = config_.rtp.ssrcs[simulcast_idx];
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vie_channel.cc | 148 // RTP/RTCP initialization. 178 // Make sure we don't get more callbacks from the RTP module. 236 StreamDataCounters rtp; local 238 GetSendStreamDataCounters(&rtp, &rtx); 239 StreamDataCounters rtp_rtx = rtp; 251 static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000)); 301 StreamDataCounters rtp; local 303 GetReceiveStreamDataCounters(&rtp, &rtx); 304 StreamDataCounters rtp_rtx = rtp; 314 static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000)) [all...] |
video_quality_test.cc | 800 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 801 video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; 803 video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); 805 video_send_config_.rtp.extensions.clear(); 807 video_send_config_.rtp.extensions.push_back( 811 video_send_config_.rtp.extensions.push_back(RtpExtension( 823 video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 824 video_receive_configs_[i].rtp.rtx[kSendRtxPayloadType].ssrc = 826 video_receive_configs_[i].rtp.rtx[kSendRtxPayloadType].payload_type = 828 video_receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe [all...] |
video_send_stream_tests.cc | 118 send_config->rtp.c_name = kCName; 154 send_config->rtp.extensions.clear(); 155 send_config->rtp.extensions.push_back(RtpExtension( 160 EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet."; 197 send_config->rtp.extensions.clear(); 198 send_config->rtp.extensions.push_back( 203 EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet."; 241 send_config->rtp.extensions.clear(); 242 send_config->rtp.extensions.push_back( 247 EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet." [all...] |
/external/webrtc/talk/media/webrtc/ |
webrtcvideoengine2_unittest.cc | 68 "urn:ietf:params:rtp-hdrext:unsupported"; 101 it->second == config.rtp.rtx.payload_type); 103 if (config.rtp.fec.red_rtx_payload_type != -1) { 104 it = rtx_types.find(config.rtp.fec.red_payload_type); 106 it->second == config.rtp.fec.red_rtx_payload_type); [all...] |
webrtcvideoengine2.cc | 405 !is_screencast && parameters_.config.rtp.ssrcs.size() == 1; [all...] |
fakewebrtccall.cc | 276 if (p->GetConfig().rtp.ssrc == ssrc) { 289 if (p->GetConfig().rtp.remote_ssrc == ssrc) { 399 if (receiver->GetConfig().rtp.remote_ssrc == ssrc) 406 if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
rtp_rtcp_impl_unittest.cc | 414 StreamDataCounters rtp; local 416 rtp.first_packet_time_ms = kStartTimeMs; 417 rtp.transmitted.packets = 1; 418 rtp.transmitted.payload_bytes = 1; 419 rtp.transmitted.header_bytes = 2; 420 rtp.transmitted.padding_bytes = 3; 421 EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes + 422 rtp.transmitted.header_bytes + 423 rtp.transmitted.padding_bytes) [all...] |
/frameworks/av/media/libstagefright/wifi-display/ |
Android.mk | 8 rtp/RTPSender.cpp \
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/external/curl/lib/ |
rtsp.c | 70 * Parse and write out any available RTP data. 72 * nread: amount of data left after k->str. will be modified if RTP 74 * readmore: whether or not the RTP parser needs more data right away 138 * The server may send us RTP data at any point, and RTSPREQ_RECEIVE does not 223 infof(data, "Got an RTP Receive with a CSeq of %ld\n", CSeq_recv); 307 /* Treat interleaved RTP as body*/ 610 char *rtp; /* moving pointer to rtp data */ local 627 rtp = rtspc->rtp_buf; 632 rtp = k->str [all...] |
/external/webrtc/webrtc/ |
audio_receive_stream.h | 66 // Receive-stream specific RTP settings. 67 struct Rtp { 82 // RTP header extensions used for the received stream. 84 } rtp; member in struct:webrtc::AudioReceiveStream::Config
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video_receive_stream.h | 39 // Received RTP packets with this payload type will be sent to this decoder 86 // Receive-stream specific RTP settings. 87 struct Rtp { 127 // Map from video RTP payload type -> RTX config. 136 // RTP header extensions used for the received stream. 138 } rtp; member in struct:webrtc::VideoReceiveStream::Config
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video_send_stream.h | 100 struct Rtp { 108 // Max RTP packet size delivered to send transport from VideoEngine. 111 // RTP header extensions to use for this send stream. 120 // Settings for RTP retransmission payload format, see RFC 4588 for 133 } rtp; member in struct:webrtc::VideoSendStream::Config
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/frameworks/opt/net/voip/src/java/android/net/rtp/ |
RtpStream.java | 17 package android.net.rtp; 26 * packets with media payloads over Real-time Transport Protocol (RTP).
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AudioCodec.java | 17 package android.net.rtp; 39 * The RTP payload type of the encoding. 100 * @param type The payload type of the encoding defined in RTP/AVP.
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AudioGroup.java | 17 package android.net.rtp;
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/external/webrtc/webrtc/call/ |
call_perf_tests.cc | 89 // RTCP sender where it sends RTCP SR before any RTP packets, which leads 90 // to a bogus NTP/RTP mapping. 109 // We need two RTCP SR reports to map between RTP and NTP. More than two 292 audio_send_config.rtp.ssrc = kAudioSendSsrc; 299 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 301 video_send_config_.rtp.fec.red_payload_type = kRedPayloadType; 302 video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; 303 video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType; 304 video_receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; 306 video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000 [all...] |
rtc_event_log.cc | 270 receiver_config->set_remote_ssrc(config.rtp.remote_ssrc); 271 receiver_config->set_local_ssrc(config.rtp.local_ssrc); 273 receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtp.rtcp_mode)); 274 receiver_config->set_remb(config.rtp.remb); 276 for (const auto& kv : config.rtp.rtx) { 283 for (const auto& e : config.rtp.extensions) { 308 for (const auto& ssrc : config.rtp.ssrcs) { 312 for (const auto& e : config.rtp.extensions) { 319 for (const auto& rtx_ssrc : config.rtp.rtx.ssrcs) { 322 sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type) [all...] |
call.cc | 309 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == 311 audio_send_ssrcs_[config.rtp.ssrc] = send_stream; 328 audio_send_stream->config().rtp.ssrc); 342 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) == 344 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; 360 audio_receive_stream->config().rtp.remote_ssrc); 390 for (uint32_t ssrc : config.rtp.ssrcs) { 445 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) == 447 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; 449 VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it [all...] |
/external/skia/src/gpu/ |
GrDrawingManager.h | 47 GrRenderTargetOpList* newOpList(GrRenderTargetProxy* rtp);
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/external/dhcpcd-6.8.2/ |
ipv4.c | 577 struct rt *rtp, *rtn; local 585 TAILQ_FOREACH(rtp, rt, next) { 586 if (rtp->dest.s_addr != INADDR_ANY) 590 if (rtn == rtp) 593 if (rtn->dest.s_addr == rtp->gate.s_addr) 596 cp = (const char *)&rtp->gate.s_addr; 607 if (rtn != rtp) 618 ifp->name, inet_ntoa(rtp->gate)); 620 rtp->gate.s_addr = 0; 629 ifp->name, inet_ntoa(rtp->gate)) [all...] |
/cts/apps/CtsVerifier/src/com/android/cts/verifier/audio/ |
SoundPlayerObject.java | 25 import android.net.rtp.AudioStream;
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