| /external/libvorbis/doc/ |
| a2-encapsulation-rtp.tex | 4 \section{Vorbis encapsulation in RTP} \label{vorbis:over:rtp}
6 % TODO: Include draft-rtp.xml somehow?
8 Please consult RFC 5215 \textit{``RTP Payload Format for Vorbis Encoded
9 Audio''} for description of how to embed Vorbis audio in an RTP stream.
|
| Makefile.am | 68 a2-encapsulation-rtp.tex \
|
| /frameworks/av/media/libstagefright/wifi-display/rtp/ |
| RTPSender.cpp | 219 uint8_t *rtp = udpPacket->data(); local
220 rtp[0] = 0x80;
221 rtp[1] = packetType;
223 rtp[2] = (mRTPSeqNo >> 8) & 0xff;
224 rtp[3] = mRTPSeqNo & 0xff;
229 rtp[4] = rtpTime >> 24;
230 rtp[5] = (rtpTime >> 16) & 0xff;
231 rtp[6] = (rtpTime >> 8) & 0xff;
232 rtp[7] = rtpTime & 0xff;
234 rtp[8] = kSourceID >> 24
264 uint8_t *rtp = udpPacket->data(); local
[all...] |
| /cts/tests/tests/net/src/android/net/rtp/cts/ |
| AudioStreamTest.java | 16 package android.net.rtp.cts;
18 import android.net.rtp.AudioCodec;
19 import android.net.rtp.AudioStream;
|
| AudioGroupTest.java | 16 package android.net.rtp.cts;
20 import android.net.rtp.AudioCodec;
21 import android.net.rtp.AudioGroup;
22 import android.net.rtp.AudioStream;
23 import android.net.rtp.RtpStream;
|
| AudioCodecTest.java | 16 package android.net.rtp.cts;
18 import android.net.rtp.AudioCodec;
|
| /external/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
79 #include "rtp.h"
119 * program_type distinguishes the [s]rtp sender and receiver cases
314 crypto_policy_set_rtp_default(&policy.rtp);
318 crypto_policy_set_aes_cm_128_null_auth(&policy.rtp);
322 crypto_policy_set_null_cipher_hmac_sha1_80(&policy.rtp);
335 policy.rtp.sec_serv = sec_servs;
370 * application is now a vanilla-flavored RTP application
[all...] |
| srtp_driver.c | 319 crypto_policy_set_rtp_default(&policy.rtp);
366 * (malloced) example RTP packet whose data field has the length given
390 hdr->version = 2; /* RTP version two */
403 /* set RTP data to 0xab */
499 len = msg_len_octets + 12; /* add in rtp header length */
580 int tag_length = policy->rtp.auth_tag_len;
660 if ((policy->rtp.sec_serv & sec_serv_conf) && (msg_len_octets >= 4)) {
713 if (policy->rtp.sec_serv & sec_serv_auth) {
779 int tag_length = policy->rtp.auth_tag_len;
859 if ((policy->rtp.sec_serv & sec_serv_conf) && (msg_len_octets >= 4))
[all...] |
| dtls_srtp_driver.c | 181 err = crypto_policy_set_from_profile_for_rtp(&policy.rtp, profile);
202 * (malloced) example RTP packet whose data field has the length given
226 hdr->version = 2; /* RTP version two */
239 /* set RTP data to 0xab */
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| 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
|
| webrtcvoe.h | 112 webrtc::VoERTP_RTCP* rtp,
125 rtp_(rtp),
140 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } function in class:cricket::VoEWrapper
|
| webrtcvideoengine.cc | 148 // Extension header for RTP timestamp offset, see RFC 5450 for details:
151 "urn:ietf:params:rtp-hdrext:toffset";
155 // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
157 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
238 // Convert 90K rtp timestamp to ns timestamp.
244 // Send the rtp timestamp to renderer as the VideoFrame timestamp.
878 // Load our RTP Header extensions.
3118 webrtc::ViERTP_RTCP* rtp = engine()->vie()->rtp(); local
[all...] |
| webrtcvoiceengine.cc | 113 "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
124 // draft-spittka-payload-rtp-opus-03
378 // Load our RTP Header extensions.
2419 webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp(); local
[all...] |
| /frameworks/opt/net/voip/src/java/android/net/rtp/ |
| AudioStream.java | 17 package android.net.rtp;
24 * Real-time Transport Protocol (RTP). Two different classes are developed in
130 * Returns the RTP payload type for dual-tone multi-frequency (DTMF) digits,
140 * Sets the RTP payload type for dual-tone multi-frequency (DTMF) digits.
143 * RTP payload type for DTMF is assigned dynamically, so it must be in the
148 * @param type The RTP payload type to be used or {@code -1} to disable it.
|
| 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;
|
| /frameworks/av/media/libstagefright/wifi-display/ |
| Android.mk | 8 rtp/RTPSender.cpp \
|
| /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/third_party/libjingle/source/talk/session/phone/ |
| srtpfilter.cc | 361 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
364 crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); // rtp is 32,
391 rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
|
| /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...] |
| /frameworks/opt/net/voip/src/java/android/net/sip/ |
| SipAudioCall.java | 21 import android.net.rtp.AudioCodec;
22 import android.net.rtp.AudioGroup;
23 import android.net.rtp.AudioStream;
24 import android.net.rtp.RtpStream;
740 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
759 && "RTP/AVP".equals(media.getProtocol())) {
770 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
823 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
[all...] |
| /sdk/eclipse/plugins/com.android.ide.eclipse.adt/src/com/android/ide/eclipse/adt/internal/refactorings/core/ |
| AndroidPackageRenameParticipant.java | 228 RenameTypeProcessor rtp = local
230 if (rtp != null) {
231 String pattern = rtp.getFilePatterns();
232 boolean updQualf = rtp.getUpdateQualifiedNames();
|
| AndroidTypeRenameParticipant.java | 206 RenameTypeProcessor rtp = local
208 if (rtp != null) {
209 String pattern = rtp.getFilePatterns();
210 boolean updQualf = rtp.getUpdateQualifiedNames();
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| srtpfilter.cc | 166 // differently in RTP/RTCP mux and non-mux modes.
168 // - In the non-muxed case, RTP and RTCP are keyed with different
577 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
580 crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); // rtp is 32,
607 rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
|
