| /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;
|
| /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(
|
| 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];
|
| 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) {
|
| /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 \
|
| /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
|
| 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
|
| 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
|
| /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).
|
| 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.
|
| AudioGroup.java | 17 package android.net.rtp;
|
| /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);
|
| /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;
|
