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Searched refs:senders (Results 1 – 23 of 23) sorted by relevance

/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/
Dbwe_test.cc164 if (!uplink_.senders().empty()) { in RunFor()
165 simulation_interval_ms_ = uplink_.senders()[0]->GetFeedbackIntervalMs(); in RunFor()
166 } else if (!downlink_.senders().empty()) { in RunFor()
167 simulation_interval_ms_ = downlink_.senders()[0]->GetFeedbackIntervalMs(); in RunFor()
285 std::vector<PacketSender*> senders; in RunFairnessTest() local
293 senders.push_back(new PacedVideoSender(&uplink_, sources.back(), bwe_type)); in RunFairnessTest()
298 senders.push_back(new TcpSender(&uplink_, tcp_flow, offsets_ms[tcp_flow])); in RunFairnessTest()
333 senders[media_flow], &link_share)); in RunFairnessTest()
346 senders[tcp_flow], &link_share)); in RunFairnessTest()
393 for (PacketSender* sender : senders) in RunFairnessTest()
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Dbwe_test.h58 const std::vector<PacketSender*>& senders() { return senders_; } in senders() function
/external/webrtc/talk/app/webrtc/java/src/org/webrtc/
DPeerConnection.java175 private List<RtpSender> senders; field in PeerConnection
182 senders = new LinkedList<RtpSender>(); in PeerConnection()
230 senders.add(new_sender); in createSender()
238 for (RtpSender sender : senders) { in getSenders()
241 senders = nativeGetSenders(); in getSenders()
242 return Collections.unmodifiableList(senders); in getSenders()
274 for (RtpSender sender : senders) { in dispose()
277 senders.clear(); in dispose()
/external/webrtc/talk/app/webrtc/
Dpeerconnectioninterface_unittest.cc331 const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders, in ContainsSender() argument
333 for (const auto& sender : senders) { in ContainsSender()
2116 auto senders = pc_->GetSenders(); in TEST_F() local
2117 EXPECT_EQ(4u, senders.size()); in TEST_F()
2118 EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0])); in TEST_F()
2119 EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0])); in TEST_F()
2120 EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1])); in TEST_F()
2121 EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1])); in TEST_F()
2129 senders = pc_->GetSenders(); in TEST_F()
2130 EXPECT_EQ(2u, senders.size()); in TEST_F()
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Dstatscollector_unittest.cc620 stats_read->senders.push_back(*voice_sender_info); in SetupAndVerifyAudioTrackStats()
844 stats_read.senders.push_back(video_sender_info); in TEST_F()
892 stats_read.senders.push_back(video_sender_info); in TEST_F()
999 stats_read.senders.push_back(video_sender_info); in TEST_F()
1063 stats_read.senders.push_back(video_sender_info); in TEST_F()
1154 stats_read.senders.push_back(video_sender_info); in TEST_F()
1578 stats_read.senders.push_back(voice_sender_info); in TEST_F()
1657 stats_read.senders.push_back(voice_sender_info); in TEST_F()
Dpeerconnection.cc368 const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders, in AddSendStreams() argument
372 for (const auto& sender : senders) { in AddSendStreams()
780 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders; in GetSenders() local
782 senders.push_back(RtpSenderProxy::Create(signaling_thread(), sender.get())); in GetSenders()
784 return senders; in GetSenders()
Dstatscollector.cc798 ExtractStatsFromList(voice_info.senders, transport_id, this, in ExtractVoiceInfo()
830 ExtractStatsFromList(video_info.senders, transport_id, this, in ExtractVideoInfo()
/external/jemalloc/test/unit/
Dmq.c68 thd_t senders[NSENDERS]; in TEST_BEGIN() local
75 thd_create(&senders[i], thd_sender_start, (void *)&mq); in TEST_BEGIN()
79 thd_join(senders[i], NULL); in TEST_BEGIN()
/external/webrtc/talk/media/base/
Dvideoengine_unittest.h505 ASSERT_EQ(1U, info.senders.size()); in GetStats()
508 EXPECT_GT(info.senders[0].bytes_sent, 0); in GetStats()
509 EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent); in GetStats()
510 EXPECT_EQ(0.0, info.senders[0].fraction_lost); in GetStats()
511 EXPECT_EQ(0, info.senders[0].firs_rcvd); in GetStats()
512 EXPECT_EQ(0, info.senders[0].plis_rcvd); in GetStats()
513 EXPECT_EQ(0, info.senders[0].nacks_rcvd); in GetStats()
514 EXPECT_EQ(DefaultCodec().width, info.senders[0].send_frame_width); in GetStats()
515 EXPECT_EQ(DefaultCodec().height, info.senders[0].send_frame_height); in GetStats()
516 EXPECT_GT(info.senders[0].framerate_input, 0); in GetStats()
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Dmediachannel.h909 senders.clear(); in Clear()
912 std::vector<VoiceSenderInfo> senders; member
918 senders.clear(); in Clear()
922 std::vector<VideoSenderInfo> senders; member
929 senders.clear(); in Clear()
932 std::vector<DataSenderInfo> senders; member
/external/webrtc/talk/media/webrtc/
Dwebrtcvideoengine2_unittest.cc2448 EXPECT_EQ(kVp8Codec.name, info.senders[0].codec_name); in TEST_F()
2460 info.senders[0].encoder_implementation_name); in TEST_F()
2472 EXPECT_EQ(stats.avg_encode_time_ms, info.senders[0].avg_encode_ms); in TEST_F()
2473 EXPECT_EQ(stats.encode_usage_percent, info.senders[0].encode_usage_percent); in TEST_F()
2489 ASSERT_EQ(1u, info.senders.size()); in TEST_F()
2490 EXPECT_EQ(123, info.senders[0].send_frame_width); in TEST_F()
2491 EXPECT_EQ(90, info.senders[0].send_frame_height); in TEST_F()
2527 ASSERT_EQ(1U, info.senders.size()); in TEST_F()
2528 EXPECT_EQ(1, info.senders[0].adapt_changes); in TEST_F()
2530 info.senders[0].adapt_reason); in TEST_F()
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Dwebrtcvoiceengine_unittest.cc2075 EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); in TEST_F()
2076 for (const auto& sender : info.senders) { in TEST_F()
2090 EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); in TEST_F()
2101 EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size()); in TEST_F()
2244 EXPECT_EQ(1u, info.senders.size()); in TEST_F()
2245 VerifyVoiceSenderInfo(info.senders[0], false); in TEST_F()
2256 VerifyVoiceSenderInfo(info.senders[0], true); in TEST_F()
2264 EXPECT_EQ(1u, info.senders.size()); in TEST_F()
2275 EXPECT_EQ(1u, info.senders.size()); in TEST_F()
Dwebrtcvideoengine2.cc1210 for (size_t i = 0; i < info->senders.size(); ++i) { in GetStats()
1211 info->senders[i].rtt_ms = stats.rtt_ms; in GetStats()
1222 video_media_info->senders.push_back(it->second->GetVideoSenderInfo()); in FillSenderStats()
Dwebrtcvoiceengine.cc2353 RTC_DCHECK(info->senders.size() == 0); in GetStats()
2374 info->senders.push_back(sinfo); in GetStats()
/external/webrtc/webrtc/modules/remote_bitrate_estimator/
Dbwe_simulations.cc248 rtc::scoped_ptr<VideoSender> senders[kNumFlows]; in TEST_P() local
254 senders[i].reset(new VideoSender(&uplink_, sources[i].get(), GetParam())); in TEST_P()
/external/srtp/
DChanges172 works with multiple srtp senders. For now, this functionality is
/external/chromium-trace/catapult/dashboard/docs/
Dadmin-tasks.md142 ### Whitelisting senders of data
/external/bzip2/
DCHANGES196 Here are the changes in 1.0.2. Bug-reporters and/or patch-senders in
/external/webrtc/talk/app/webrtc/java/jni/
Dpeerconnection_jni.cc1814 auto senders = ExtractNativePC(jni, j_pc)->GetSenders(); in JOW() local
1815 for (const auto& sender : senders) { in JOW()
/external/srtp/doc/
Dintro.txt251 feedback from receivers to senders. An @e SRTP @e session is
Drfc3711.txt270 feedback to RTP senders, or maintain packet sequence counters. SRTCP
1876 SRTP senders SHALL count the amount of SRTP and SRTCP traffic being
2056 scenarios where for example, there are multiple senders that can
/external/iproute2/doc/
Dip-cref.tex1196 The local senders get an \verb|EHOSTUNREACH| error.
1198 are discarded silently. The local senders get an \verb|EINVAL| error.
1201 prohibited\/} is generated. The local senders get an \verb|EACCES| error.
1211 is generated. The local senders get an \verb|ENETUNREACH| error.
1704 returned to local senders when they try to use this route.
1706 senders, according to the rules described above in the subsection
/external/libmicrohttpd/src/datadir/
Dspdy-draft.txt875 senders of FRAME_TOO_LARGE MUST close the session.