| /external/bluetooth/bluez/audio/ |
| gstrtpsbcpay.h | 25 #include <gst/rtp/gstbasertppayload.h>
27 #include <gst/rtp/gstrtpbuffer.h>
|
| gsta2dpsink.h | 28 #include <gst/rtp/gstbasertppayload.h>
50 GstBaseRTPPayload *rtp; member in struct:_GstA2dpSink
|
| gsta2dpsink.c | 268 if (self->rtp) {
269 GST_LOG_OBJECT(self, "removing rtp element from the bin");
270 if (!gst_bin_remove(GST_BIN(self), GST_ELEMENT(self->rtp)))
271 GST_WARNING_OBJECT(self, "failed to remove rtp "
274 self->rtp = NULL;
456 if (!gst_element_link(GST_ELEMENT(self->rtp), sink)) {
487 /* if we already have a rtp, we don't need a new one */
488 if (self->rtp != NULL)
491 rtppay = gst_a2dp_sink_init_element(self, "rtpsbcpay", "rtp",
496 self->rtp = GST_BASE_RTP_PAYLOAD(rtppay)
[all...] |
| gstavdtpsink.c | 38 #include <gst/rtp/gstrtpbuffer.h>
43 #include "rtp.h"
105 GST_STATIC_CAPS("application/x-rtp, "
112 "application/x-rtp, "
117 "application/x-rtp, "
[all...] |
| /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 */
|
| /frameworks/base/voip/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;
|
| /external/dhcpcd/ |
| configure.c | 590 struct rt *rtp, *rtl, *rtn; local
593 for (rtp = rt, rtl = NULL; rtp; rtl = rtp, rtp = rtp->next) {
594 if (rtp->dest.s_addr != INADDR_ANY)
597 for (rtn = rt; rtn != rtp; rtn = rtn->next) {
599 if (rtn->dest.s_addr == rtp->gate.s_addr)
602 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/base/voip/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;
731 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
749 && "RTP/AVP".equals(media.getProtocol())) {
760 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
809 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
[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/srtp/ |
| Makefile | 1 # Makefile for secure rtp
129 test/rtpw$(EXE): test/rtpw.c test/rtp.c
|
| /frameworks/base/telephony/java/com/android/internal/telephony/sip/ |
| SipPhone.java | 21 import android.net.rtp.AudioGroup;
[all...] |
| /prebuilt/sdk/12/ |
| android.jar | |
| /prebuilt/sdk/14/ |
| android.jar | |
