| /external/srtp/test/ |
| 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/libsrtp/srtp/test/ |
| srtp_driver.c | 343 crypto_policy_set_rtp_default(&policy.rtp);
396 * (malloced) example RTP packet whose data field has the length given
420 hdr->version = 2; /* RTP version two */
433 /* set RTP data to 0xab */
528 len = msg_len_octets + 12; /* add in rtp header length */
624 int tag_length = policy->rtp.auth_tag_len;
704 if ((policy->rtp.sec_serv & sec_serv_conf) && (msg_len_octets >= 4)) {
761 if (policy->rtp.sec_serv & sec_serv_auth) {
830 int tag_length = policy->rtp.auth_tag_len;
910 if ((policy->rtp.sec_serv & sec_serv_conf) && (msg_len_octets >= 4))
[all...] |
| dtls_srtp_driver.c | 188 err = crypto_policy_set_from_profile_for_rtp(&policy.rtp, profile);
215 * (malloced) example RTP packet whose data field has the length given
239 hdr->version = 2; /* RTP version two */
252 /* set RTP data to 0xab */
|
| /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/media/cast/audio_sender/ |
| audio_sender.cc | 83 transport_config.rtp.config = audio_config.rtp_config;
86 transport_config.rtp.max_outstanding_frames = max_unacked_frames_;
|
| /external/chromium_org/media/cast/video_sender/ |
| video_sender.cc | 80 transport_config.rtp.config = video_config.rtp_config;
81 transport_config.rtp.max_outstanding_frames = max_unacked_frames_;
|
| /external/libvorbis/doc/ |
| Makefile.am | 68 a2-encapsulation-rtp.tex \
|
| Vorbis_I_spec.tex | 117 \include{a2-encapsulation-rtp}
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvoiceengine.cc | 118 // draft-spittka-payload-rtp-opus-03
383 // Load our RTP Header extensions.
2652 webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp(); local
[all...] |
| /external/chromium_org/third_party/libsrtp/srtp/srtp/ |
| srtp.c | 77 /* Check RTP header length */
105 * This function allocates the stream context, rtp and rtcp ciphers
119 stat = crypto_kernel_alloc_cipher(p->rtp.cipher_type,
121 p->rtp.cipher_key_len);
128 stat = crypto_kernel_alloc_auth(p->rtp.auth_type,
130 p->rtp.auth_key_len,
131 p->rtp.auth_tag_len);
445 /* If RTP or RTCP have a key length > AES-128, assume matching kdf. */
640 srtp->rtp_services = p->rtp.sec_serv;
757 /* Verify RTP header *
[all...] |
| /external/srtp/srtp/ |
| srtp.c | 84 * This function allocates the stream context, rtp and rtcp ciphers
98 stat = crypto_kernel_alloc_cipher(p->rtp.cipher_type,
100 p->rtp.cipher_key_len);
106 stat = crypto_kernel_alloc_auth(p->rtp.auth_type,
108 p->rtp.auth_key_len,
109 p->rtp.auth_tag_len);
497 srtp->rtp_services = p->rtp.sec_serv;
688 * encrypted - the encrypted portion starts after the rtp header
957 * decrypted - the encrypted portion starts after the rtp header
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| srtpfilter.cc | 173 // differently in RTP/RTCP mux and non-mux modes.
175 // - In the non-muxed case, RTP and RTCP are keyed with different
648 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
651 crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); // rtp is 32,
672 // We want to set this option only for rtp packets.
677 policy.rtp.auth_type = EXTERNAL_HMAC_SHA1;
689 rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
|
| channel_unittest.cc | 472 // Set SSRC in the rtp packet copy.
1817 TransportChannel* rtp = channel1_->transport_channel(); local
1849 TransportChannel* rtp = channel1_->transport_channel(); local
[all...] |
| /external/srtp/ |
| Makefile | 1 # Makefile for secure rtp
129 test/rtpw$(EXE): test/rtpw.c test/rtp.c
|
| /external/chromium_org/media/cast/rtcp/ |
| rtcp_unittest.cc | 40 // Packet lists imply a RTP packet.
178 config.rtp.config.ssrc = kSenderSsrc;
179 config.rtp.max_outstanding_frames = 1;
|
| /external/chromium_org/third_party/libsrtp/srtp/ |
| Makefile | 1 # Makefile for secure rtp
132 test/rtpw$(EXE): test/rtpw.c test/rtp.c test/getopt_s.c
|
| /frameworks/opt/telephony/src/java/com/android/internal/telephony/sip/ |
| SipPhone.java | 21 import android.net.rtp.AudioGroup;
[all...] |
| /frameworks/base/ |
| Android.mk | [all...] |
| /external/robolectric/lib/main/ |
| android.jar | |
| /prebuilts/sdk/12/ |
| android.jar | |
| /prebuilts/sdk/14/ |
| android.jar | |
| /prebuilts/sdk/18/ |
| android.jar | |
