| /frameworks/av/media/libstagefright/wifi-display/ |
| Android.mk | 8 rtp/RTPSender.cpp \
|
| /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;
|
| /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
|
| 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.
|
| /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
|
| 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
|
| /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/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;
|
| /external/chromium_org/chrome/common/ |
| cast_messages.h | 62 IPC_STRUCT_TRAITS_MEMBER(rtp)
69 IPC_STRUCT_TRAITS_MEMBER(rtp)
|
| /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(
|
| 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.
|
| /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...] |
