/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_remb_unittest.cc | 45 MockRtpRtcp rtp; local 46 vie_remb_->AddReceiveChannel(&rtp); 47 vie_remb_->AddRembSender(&rtp); 56 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _)) 61 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, 1, _)) 65 vie_remb_->RemoveReceiveChannel(&rtp); 66 vie_remb_->RemoveRembSender(&rtp); 70 MockRtpRtcp rtp; local 71 vie_remb_->AddReceiveChannel(&rtp); 72 vie_remb_->AddRembSender(&rtp); 198 MockRtpRtcp rtp; local 229 MockRtpRtcp rtp; local [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | mt_test_common.h | 28 SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp, argument 31 _rtp(rtp), 49 // constructor input: (receive side) rtp module to send encoded data to 63 SharedRTPState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp) : argument 65 _rtp(rtp) {} 74 SharedTransportState(webrtc::RtpRtcp& rtp, TransportCallback& transport): argument 75 _rtp(rtp),
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H A D | mt_rx_tx_test.cc | 152 new RTPSendCompleteCallback(Clock::GetRealTimeClock(), "dump.rtp"); 158 RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(configuration); local 192 TEST(rtp->RegisterSendPayload(video_codec) == 0); 195 TEST(rtp->SetGenericFECStatus(fecEnabled, VCM_RED_PAYLOAD_TYPE, 235 PacketRequester packetRequester(*rtp); 238 VCMRTPEncodeCompleteCallback* encodeCompleteCallback = new VCMRTPEncodeCompleteCallback(rtp); 259 rtp->SetFecParameters(&delta_params, &key_params); 265 SharedRTPState mtState(*vcm, *rtp); // receive side 266 SendSharedState mtSendState(*vcm, *rtp, args); // send side 373 delete rtp; [all...] |
H A D | test_callbacks.h | 96 VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) : argument 99 _RTPModule(rtp) {} 161 // Constructor input: (receive side) rtp module to send encoded data to 210 PacketRequester(RtpRtcp& rtp) : argument 211 _rtp(rtp) {} 247 void RegisterRtpModule(RtpRtcp* rtp) {_rtp = rtp;} argument
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/external/chromium_org/media/cast/logging/ |
H A D | receiver_time_offset_estimator_impl.cc | 20 uint32 rtp, 24 uint64 key = (static_cast<uint64>(rtp) << 32) | (packet_id << 1) | 31 uint32 rtp, 35 uint64 key = (static_cast<uint64>(rtp) << 32) | (packet_id << 1) | 19 SetSent( uint32 rtp, uint32 packet_id, bool audio, base::TimeTicks t) argument 30 SetReceived( uint32 rtp, uint16 packet_id, bool audio, base::TimeTicks t) argument
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/external/chromium_org/third_party/webrtc/ |
H A D | video_send_stream.h | 117 } rtp; member in struct:webrtc::VideoSendStream::Config
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H A D | video_receive_stream.h | 139 } rtp; member in struct:webrtc::VideoReceiveStream::Config
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcvie.h | 95 webrtc::ViERender* render, webrtc::ViERTP_RTCP* rtp, 104 rtp_(rtp), 116 webrtc::ViERTP_RTCP* rtp() { return rtp_.get(); } function in class:cricket::ViEWrapper 93 ViEWrapper(webrtc::ViEBase* base, webrtc::ViECodec* codec, webrtc::ViECapture* capture, webrtc::ViENetwork* network, webrtc::ViERender* render, webrtc::ViERTP_RTCP* rtp, webrtc::ViEImageProcess* image, webrtc::ViEExternalCodec* ext_codec) argument
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H A D | webrtcvoe.h | 112 webrtc::VoERTP_RTCP* rtp, 125 rtp_(rtp), 140 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } function in class:cricket::VoEWrapper 103 VoEWrapper(webrtc::VoEAudioProcessing* processing, webrtc::VoEBase* base, webrtc::VoECodec* codec, webrtc::VoEDtmf* dtmf, webrtc::VoEFile* file, webrtc::VoEHardware* hw, webrtc::VoEExternalMedia* media, webrtc::VoENetEqStats* neteq, webrtc::VoENetwork* network, webrtc::VoERTP_RTCP* rtp, webrtc::VoEVideoSync* sync, webrtc::VoEVolumeControl* volume) argument
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H A D | webrtcvoiceengine.cc | 119 // draft-spittka-payload-rtp-opus-03 2032 engine()->voe()->rtp()->SetNACKStatus(channel, false, 0); 2034 engine()->voe()->rtp()->SetREDStatus(channel, false); 2038 engine()->voe()->rtp()->SetFECStatus(channel, false); 2086 if (engine()->voe()->rtp()->SetREDStatus(channel, true, it->id) == -1) { 2091 if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) { 2265 engine()->voe()->rtp()->SetNACKStatus(channel, true, kNackMaxPackets); 2268 engine()->voe()->rtp()->SetNACKStatus(channel, false, 0); 2310 // The default channel may or may not be in |receive_channels_|. Set the rtp 2355 // The default channel may or may not be in |send_channels_|. Set the rtp 2704 webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp(); local [all...] |
/external/chromium_org/third_party/libsrtp/srtp/include/ |
H A D | srtp.h | 216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member in struct:srtp_policy_t 319 * - err_status_replay_fail rtp sequence number was non-increasing 350 * rtp packet if err_status_ok was returned; otherwise, the value of 355 * complete rtp packet after the call, if err_status_ok was returned. 867 * rtp packet if err_status_ok was returned; otherwise, the value of 872 * and of the complete rtp packet after the call, if err_status_ok was
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/external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/ |
H A D | voice_engine_jni.cc | 64 rtp(webrtc::VoERTP_RTCP::GetInterface(ve)) { 73 CHECK(rtp != NULL, "Failed to acquire rtp interface"); 87 ReleaseSubApi(rtp); 125 webrtc::VoERTP_RTCP* const rtp; member in class:__anon15741::VoiceEngineData::webrtc 406 return voe_data->rtp->StartRTPDump( 414 return voe_data->rtp->StopRTPDump(
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H A D | video_engine_jni.cc | 159 rtp(webrtc::ViERTP_RTCP::GetInterface(vie)), 167 CHECK(rtp != NULL, "Failed to acquire rtp interface"); 183 ReleaseSubApi(rtp); 260 webrtc::ViERTP_RTCP* const rtp; member in class:__anon15740::VideoEngineData::webrtc 580 return vie_data->rtp->SetNACKStatus(channel, enable); 587 return vie_data->rtp->SetKeyFrameRequestMethod( 599 if (vie_data->rtp->GetReceivedRTCPStatistics(channel, fraction_lost, 643 return vie_data->rtp->StartRTPDump( 650 return vie_data->rtp [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
H A D | generic_encoder.cc | 20 // Map information from info into rtp. If no relevant information is found 21 // in info, rtp is set to NULL. 22 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader** rtp) { argument 24 *rtp = NULL; 29 (*rtp)->codec = kRtpVideoVp8; 30 (*rtp)->codecHeader.VP8.InitRTPVideoHeaderVP8(); 31 (*rtp)->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId; 32 (*rtp)->codecHeader.VP8.nonReference = 34 (*rtp)->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx; 35 (*rtp) [all...] |
/external/srtp/include/ |
H A D | srtp.h | 216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member in struct:srtp_policy_t 310 * - err_status_replay_fail rtp sequence number was non-increasing 341 * rtp packet if err_status_ok was returned; otherwise, the value of 346 * complete rtp packet after the call, if err_status_ok was returned. 794 * rtp packet if err_status_ok was returned; otherwise, the value of 799 * and of the complete rtp packet after the call, if err_status_ok was
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/external/dhcpcd/ |
H A D | configure.c | 642 struct rt *rtp, *rtl, *rtn; local 645 for (rtp = rt, rtl = NULL; rtp; rtl = rtp, rtp = rtp->next) { 646 if (rtp->dest.s_addr != INADDR_ANY) 649 for (rtn = rt; rtn != rtp; rtn = rtn->next) { 651 if (rtn->dest.s_addr == rtp->gate.s_addr) 654 cp = (const char *)&rtp [all...] |
/external/chromium_org/content/browser/renderer_host/p2p/ |
H A D | socket_host.cc | 70 // Verifies rtp header and message length. 71 bool ValidateRtpHeader(const char* rtp, size_t length, size_t* header_length) { argument 80 size_t cc_count = rtp[0] & 0x0F; 88 if (!(rtp[0] & 0x10)) { 95 rtp += header_length_without_extension; 103 uint16 extension_length_in_32bits = rtc::GetBE16(rtp + 2); 162 void UpdateRtpAuthTag(char* rtp, argument 192 char* auth_tag = rtp + (length - tag_length); 197 // Copy ROC after end of rtp packet. 203 if (!hmac.Sign(base::StringPiece(rtp, auth_required_lengt 373 UpdateRtpAbsSendTimeExtension(char* rtp, size_t length, int extension_id, uint32 abs_send_time) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_unittest.cc | 309 void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int* out_len) { argument 311 while ((sim_clock_ >= rtp->time()) && 312 (rtp->dataLen() >= 0)) { 313 if (rtp->dataLen() > 0) { 315 rtp->parseHeader(&rtpInfo); 318 rtp->payload(), 319 rtp->payloadLen(), 320 rtp->time() * (output_sample_rate_ / 1000))); 323 ASSERT_NE(-1, rtp->readFromFile(rtp_fp_)); 364 NETEQTEST_RTPpacket rtp; local [all...] |
/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | channel_unittest.cc | 430 // Set SSRC in the rtp packet copy. 1775 TransportChannel* rtp = channel1_->transport_channel(); local 1778 rtp->SignalReadyToSend(rtp); 1781 // MediaChannel::OnReadyToSend only be called when both rtp and rtcp 1785 // rtp channel becomes not ready to send will be propagated to mediachannel 1786 channel1_->SetReadyToSend(rtp, false); 1788 channel1_->SetReadyToSend(rtp, true); 1807 TransportChannel* rtp = channel1_->transport_channel(); local 1809 // In the case of rtcp mux, the SignalReadyToSend() from rtp channe [all...] |
/external/robolectric/lib/main/ |
H A D | android.jar | META-INF/ META-INF/MANIFEST.MF com/ com/android/ com/android/internal/ com/android/internal/util/ ... |