/frameworks/av/media/libstagefright/wifi-display/ |
Android.mk | 8 rtp/RTPSender.cpp \
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/external/chromium_org/third_party/libsrtp/srtp/test/ |
rtpw.c | 4 * rtp word sender/receiver 9 * This app is a simple RTP application intended only for testing 12 * each USEC_RATE microseconds. Secure RTP protections can be 83 #include "rtp.h" 135 * program_type distinguishes the [s]rtp sender and receiver cases 334 crypto_policy_set_rtp_default(&policy.rtp); 338 crypto_policy_set_aes_cm_128_null_auth(&policy.rtp); 342 crypto_policy_set_null_cipher_hmac_sha1_80(&policy.rtp); 356 policy.rtp.sec_serv = sec_servs; 391 * application is now a vanilla-flavored RTP application [all...] |
/external/chromium_org/third_party/webrtc/video/ |
call_perf_tests.cc | 58 send_config_.rtp.ssrcs.push_back(kSendSsrc); 126 // RTCP sender where it sends RTCP SR before any RTP packets, which leads 127 // to a bogus NTP/RTP mapping. 146 // We need two RTCP SR reports to map between RTP and NTP. More than two 307 receive_config.rtp.remote_ssrc = send_config_.rtp.ssrcs[0]; 308 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; 396 // to rtp timestamp in the sender side. So here we convert the estimated 420 // Calculate the rtp timestamp offset in order to calculate the real 494 receive_config.rtp.remote_ssrc = send_config_.rtp.ssrcs[0] [all...] |
call_tests.cc | 75 send_config_.rtp.ssrcs.push_back(kSendSsrc); 89 receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0]; 90 receive_config_.rtp.local_ssrc = kReceiverLocalSsrc; 423 send_config_.rtp.nack.rtp_history_ms = rtp_history_ms; 424 receive_config_.rtp.nack.rtp_history_ms = rtp_history_ms; 522 // receive_config_.rtp.nack.rtp_history_ms = rtp_history_ms; 523 // send_config_.rtp.nack.rtp_history_ms = rtp_history_ms; 524 send_config_.rtp.fec.red_payload_type = kRedPayloadType; 525 send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType [all...] |
rampup_tests.cc | 71 // state of the RTP module we need a full module and receive statistics to 465 send_config.rtp.nack.rtp_history_ms = 1000; 466 send_config.rtp.ssrcs = ssrcs; 468 send_config.rtp.rtx.payload_type = 96; 469 send_config.rtp.rtx.ssrcs = rtx_ssrcs; 470 send_config.rtp.rtx.pad_with_redundant_payloads = true; 472 send_config.rtp.extensions.push_back( 535 send_config.rtp.nack.rtp_history_ms = 1000; 536 send_config.rtp.ssrcs.insert( 537 send_config.rtp.ssrcs.begin(), ssrcs.begin(), ssrcs.end()) [all...] |
video_send_stream_tests.cc | 74 send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]); 265 send_config_.rtp.c_name = kCName; 297 send_config_.rtp.extensions.push_back( 345 send_config_.rtp.extensions.push_back( 498 send_config_.rtp.fec.red_payload_type = kRedPayloadType; 499 send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; 573 send_config_.rtp.nack.rtp_history_ms = 1000; 574 send_config_.rtp.rtx.payload_type = retransmit_payload_type; 576 send_config_.rtp.rtx.ssrcs.push_back(retransmit_ssrc); 762 send_config_.rtp.fec.red_payload_type = kRedPayloadType [all...] |
full_stack.cc | 398 send_config.rtp.ssrcs.push_back(kSendSsrc); 432 receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; 433 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
webrtcvoe.h | 112 webrtc::VoERTP_RTCP* rtp, 125 rtp_(rtp), 140 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } function in class:cricket::VoEWrapper
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webrtcvideoengine2.cc | 576 // Convert 90K rtp timestamp to ns timestamp. 894 config->rtp.ssrcs = sp.ssrcs; 932 config->rtp.rtx.ssrcs = rtx_ssrcs; 933 config->rtp.ssrcs = ssrcs; 976 codec_settings.codec, options_, config.rtp.ssrcs.size()); 985 config.rtp.c_name = sp.cname; 986 config.rtp.fec = codec_settings.fec; 987 if (!config.rtp.rtx.ssrcs.empty()) { 988 config.rtp.rtx.payload_type = codec_settings.rtx_payload_type; 991 config.rtp.extensions = send_rtp_extensions_ [all...] |
webrtcvideoengine.cc | 3486 webrtc::ViERTP_RTCP* rtp = engine()->vie()->rtp(); local [all...] |
/external/chromium_org/media/cast/transport/ |
cast_transport_config.h | 64 CastTransportRtpConfig rtp; member in struct:media::cast::transport::CastTransportAudioConfig 74 CastTransportRtpConfig rtp; member in struct:media::cast::transport::CastTransportVideoConfig 127 // The stream timestamp, on the timeline of the signal data. For example, RTP 139 // expected to drift with respect to the elapsed time implied by the RTP 180 uint32 media_ssrc; // SSRC of the RTP packet sender.
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/external/dhcpcd/ |
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->gate.s_addr [all...] |
/external/chromium_org/third_party/webrtc/ |
video_receive_stream.h | 42 // Received RTP packets with this payload type will be sent to this decoder 87 // Receive-stream specific RTP settings. 88 struct Rtp { 89 Rtp() 133 // Map from video RTP payload type -> RTX config. 137 // RTP header extensions used for the received stream. 139 } rtp; member in struct:webrtc::VideoReceiveStream::Config
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video_send_stream.h | 80 struct Rtp { 81 Rtp() 88 // Max RTP packet size delivered to send transport from VideoEngine. 96 // RTP header extensions to use for this send stream. 105 // Settings for RTP retransmission payload format, see RFC 4588 for 123 } 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/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
neteq_unittest.cc | 310 void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int* out_len) { 312 while ((sim_clock_ >= rtp->time()) && 313 (rtp->dataLen() >= 0)) { 314 if (rtp->dataLen() > 0) { 316 rtp->parseHeader(&rtpInfo); 319 rtp->payload(), 320 rtp->payloadLen(), 321 rtp->time() * (output_sample_rate_ / 1000))); 324 ASSERT_NE(-1, rtp->readFromFile(rtp_fp_)); 351 NETEQTEST_RTPpacket rtp; local 382 NETEQTEST_RTPpacket rtp; local [all...] |
/external/chromium_org/chrome/browser/media/ |
cast_transport_host_filter_unittest.cc | 86 audio_config.rtp.max_outstanding_frames = 10; 91 video_config.rtp.max_outstanding_frames = 10;
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/external/chromium_org/chrome/common/ |
cast_messages.h | 62 IPC_STRUCT_TRAITS_MEMBER(rtp) 69 IPC_STRUCT_TRAITS_MEMBER(rtp)
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/external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/ |
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:__anon19652::VoiceEngineData::webrtc 406 return voe_data->rtp->StartRTPDump( 414 return voe_data->rtp->StopRTPDump(
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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:__anon19651::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->StopRTPDump [all...] |
/external/chromium_org/content/browser/renderer_host/p2p/ |
socket_host.h | 32 // This method can handle only RTP packet, otherwise this method must not be 34 // if present with current time and 2. update HMAC in RTP packet. 40 // Helper method which finds RTP ofset and length if the packet is encapsulated 47 CONTENT_EXPORT bool UpdateRtpAbsSendTimeExtn(char* rtp, int length, 132 // Calls |packet_dump_callback_| to record the RTP header.
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/external/chromium_org/third_party/libsrtp/srtp/include/ |
srtp.h | 56 * @defgroup SRTP Secure RTP 58 * @brief libSRTP provides functions for protecting RTP and RTCP. See 84 * the maximum number of octets that will be added to an RTP packet by 216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member in struct:srtp_policy_t 227 * transmissions must have the same RTP 243 * An SRTP session consists of all of the traffic sent to the RTP and 244 * RTCP destination transport addresses, using the RTP/SAVP (Secure 288 * @brief srtp_protect() is the Secure RTP sender-side packet processing 292 * protection to the RTP packet rtp_hdr (which has length *len_ptr) using 298 * The sequence numbers of the RTP packets presented to this functio [all...] |
/external/srtp/include/ |
srtp.h | 56 * @defgroup SRTP Secure RTP 58 * @brief libSRTP provides functions for protecting RTP and RTCP. See 84 * the maximum number of octets that will be added to an RTP packet by 216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member in struct:srtp_policy_t 227 * transmissions must have the same RTP 243 * An SRTP session consists of all of the traffic sent to the RTP and 244 * RTCP destination transport addresses, using the RTP/SAVP (Secure 279 * @brief srtp_protect() is the Secure RTP sender-side packet processing 283 * protection to the RTP packet rtp_hdr (which has length *len_ptr) using 289 * The sequence numbers of the RTP packets presented to this functio [all...] |