/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 */
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/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 */
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/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();
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AndroidTypeRenameParticipant.java | 206 RenameTypeProcessor rtp = local 208 if (rtp != null) { 209 String pattern = rtp.getFilePatterns(); 210 boolean updQualf = rtp.getUpdateQualifiedNames();
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/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_;
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/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_;
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/external/libvorbis/doc/ |
Makefile.am | 68 a2-encapsulation-rtp.tex \
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Vorbis_I_spec.tex | 117 \include{a2-encapsulation-rtp}
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/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;
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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$(EXE): test/rtpw.c test/rtp.c
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/external/chromium_org/media/cast/rtcp/ |
rtcp_unittest.cc | 40 // Packet lists imply a RTP packet. 178 config.rtp.config.ssrc = kSenderSsrc; 179 config.rtp.max_outstanding_frames = 1;
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/external/chromium_org/third_party/libsrtp/srtp/ |
Makefile | 1 # Makefile for secure rtp 132 test/rtpw$(EXE): test/rtpw.c test/rtp.c test/getopt_s.c
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/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 | |