Searched defs:rtp (Results 1 - 20 of 20) sorted by relevance

/external/chromium_org/third_party/webrtc/video_engine/
H A Dvie_remb_unittest.cc45 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
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dmt_test_common.h28 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),
H A Dmt_rx_tx_test.cc152 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;
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H A Dtest_callbacks.h96 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
/external/chromium_org/media/cast/logging/
H A Dreceiver_time_offset_estimator_impl.cc20 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
/external/chromium_org/third_party/webrtc/
H A Dvideo_send_stream.h117 } rtp; member in struct:webrtc::VideoSendStream::Config
H A Dvideo_receive_stream.h139 } rtp; member in struct:webrtc::VideoReceiveStream::Config
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
H A Dwebrtcvie.h95 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
H A Dwebrtcvoe.h112 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
H A Dwebrtcvoiceengine.cc119 // 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
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/external/chromium_org/third_party/libsrtp/srtp/include/
H A Dsrtp.h216 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
/external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/
H A Dvoice_engine_jni.cc64 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(
H A Dvideo_engine_jni.cc159 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
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/
H A Dgeneric_encoder.cc20 // 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)
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/external/srtp/include/
H A Dsrtp.h216 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
/external/dhcpcd/
H A Dconfigure.c642 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
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/external/chromium_org/content/browser/renderer_host/p2p/
H A Dsocket_host.cc70 // 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Dneteq_unittest.cc309 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
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/external/chromium_org/third_party/libjingle/source/talk/session/media/
H A Dchannel_unittest.cc430 // 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
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/external/robolectric/lib/main/
H A Dandroid.jarMETA-INF/ META-INF/MANIFEST.MF com/ com/android/ com/android/internal/ com/android/internal/util/ ...

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