| /external/smack/src/org/jivesoftware/smack/packet/ |
| Authentication.java | 21 package org.jivesoftware.smack.packet;
26 * Authentication packet, which can be used to login to a XMPP server as well
37 * Create a new authentication packet. By default, the packet will be in
40 * modes back from the server, change the type of the IQ packet to "get":
|
| RosterPacket.java | 21 package org.jivesoftware.smack.packet;
42 * Adds a roster item to the packet.
61 * Returns the number of roster items in this roster packet.
72 * Returns an unmodifiable collection for the roster items in the packet.
74 * @return an unmodifiable collection for the roster items in the packet.
|
| /external/smack/src/org/jivesoftware/smack/provider/ |
| PrivacyProvider.java | 20 import org.jivesoftware.smack.packet.DefaultPacketExtension;
21 import org.jivesoftware.smack.packet.IQ;
22 import org.jivesoftware.smack.packet.Privacy;
23 import org.jivesoftware.smack.packet.PrivacyItem;
32 * of {@link PrivacyItem} and {@link org.jivesoftware.smack.packet.PrivacyItem.PrivacyRule}.
|
| /external/smack/src/org/jivesoftware/smackx/provider/ |
| AdHocCommandDataProvider.java | 23 import org.jivesoftware.smack.packet.IQ;
24 import org.jivesoftware.smack.packet.PacketExtension;
25 import org.jivesoftware.smack.packet.XMPPError;
32 import org.jivesoftware.smackx.packet.AdHocCommandData;
33 import org.jivesoftware.smackx.packet.DataForm;
|
| /external/tcpdump/ |
| gmpls.c | 44 { 1, "Packet-Switch Capable-1"},
45 { 2, "Packet-Switch Capable-2"},
46 { 3, "Packet-Switch Capable-3"},
47 { 4, "Packet-Switch Capable-4"},
57 { 1, "Packet"},
|
| print-token.c | 155 /* Frame Control field determines interpretation of packet */
160 /* ether_type not known, print raw packet */
173 /* Some kinds of TR packet we cannot handle intelligently */
186 * to the TR header of the packet, 'h->ts' is the timestamp,
187 * 'h->len' is the length of the packet off the wire, and 'h->caplen'
|
| /system/core/fastboot/ |
| fastboot_protocol.txt | 15 * Max packet size must be 64 bytes for full-speed and 512 bytes for
25 packet no greater than 64 bytes.
27 2. Client response with a single packet no greater than 64 bytes.
43 A DATA response packet will be 12 bytes long, in the form of
53 4. Client responds with a single packet no greater than 64 bytes.
|
| /external/chromium_org/third_party/libjingle/source/talk/base/ |
| virtualsocketserver.cc | 77 class Packet : public MessageData {
79 Packet(const char* data, size_t size, const SocketAddress& from)
86 virtual ~Packet() {
270 // If we don't have a packet, then either error or wait for one to arrive.
283 // Return the packet at the front of the queue.
284 Packet* packet = recv_buffer_.front(); local
285 size_t data_read = _min(cb, packet->size());
286 std::memcpy(pv, packet->data(), data_read);
287 *paddr = packet->from()
389 Packet* packet = static_cast<Packet*>(pmsg->pdata); local
[all...] |
| /external/dnsmasq/src/ |
| forward.c | 27 /* Send a UDP packet with its source address set as "source"
29 static void send_from(int fd, int nowild, char *packet, size_t len,
47 iov[0].iov_base = packet;
279 packet straight away (helps modem users when offline) */
332 /* Keep info in case we want to re-send this packet */
387 /* If upstream is advertising a larger UDP packet size
453 sections of the packet. Find the new length here and put back pseudoheader
461 /* packet from peer server, extract data for cache, and send to
467 ssize_t n = recvfrom(fd, daemon->packet, daemon->edns_pktsz, 0, &serveraddr.sa, &addrlen);
471 /* packet buffer overwritten *
729 unsigned char *packet = whine_malloc(65536 + MAXDNAME + RRFIXEDSZ); local
[all...] |
| /external/srtp/doc/ |
| rfc3711.txt | 51 3.3. SRTP Packet Processing . . . . . . . . . . . . . . . . . 11
52 3.3.1. Packet Index Determination, and ROC, s_l Update. 13
141 SRTP can achieve high throughput and low packet expansion. SRTP
216 * limited packet expansion to support the bandwidth economy goal,
219 layers used by RTP, in particular high tolerance to packet loss
267 equivalent SRTP packet on the sending side, and intercepts SRTP
268 packets and passes an equivalent RTP packet up the stack on the
274 feedback to RTP senders, or maintain packet sequence counters. SRTCP
294 The format of an SRTP packet is illustrated in Figure 1.
321 Figure 1. The format of an SRTP packet. *Encrypted Portion is th
[all...] |
| /external/chromium_org/third_party/opus/src/doc/ |
| draft-ietf-codec-oggopus.xml | 101 A single page can contain up to 65,025 octets of packet data from up to 255
106 into packets, allowing a demuxer to recover the packet boundaries without
108 A packet is said to 'complete' on a page when the page contains the final
109 lacing value corresponding to that packet.
112 This encapsulation defines the required contents of the packet data, including
139 <section anchor="packet_organization" title="Packet Organization">
148 The first packet in the logical Ogg bitstream MUST contain the identification
151 It MUST be placed alone (without any other packet data) on the first page of
156 The second packet in the logical Ogg bitstream MUST contain the comment header,
161 However many pages it spans, the comment header packet MUST finish the page o
[all...] |
| /external/libvorbis/doc/ |
| rfc5215.xml | 91 codebooks as well as indicate if the following packet contains fragmented
175 The sequence number increments by one for each RTP data packet sent, and may be
176 used by the receiver to detect packet loss and to restore the packet sequence. This
184 Vorbis packet in the RTP payload. The clock frequency MUST be set to the sample
239 This field specifies the kind of Vorbis data stored in this RTP packet. There
240 are currently three different types of Vorbis payloads. Each packet MUST contain only a single type of Vorbis packet (e.g., you must not aggregate configuration and comment packets in the same RTP payload).
265 likely define a practical limit. Typical Vorbis packet sizes range from very
269 RTP context, to avoid fragmentation, the Vorbis data packet size SHOULD be kep
[all...] |
| /external/chromium_org/crypto/ |
| openpgp_symmetric_encryption.cc | 173 // These constants are the tag numbers for the various packet types that we
233 // ParsePacket parses an OpenPGP packet from reader. See RFC 4880, section
247 // Old format packet.
269 // New format packet.
503 // ParseSymmetricallyEncrypted parses a Symmetrically Encrypted packet. See
569 // packet. This packet is specified such that the header is always
595 // ParseLiteralData parses a Literal Data packet. See RFC 4880, section 5.9.
631 // MakePacket returns an OpenPGP packet tagged as type |tag|. It always uses
656 // SerializeLiteralData returns a Literal Data packet containing |contents
709 ByteString packet; local
[all...] |
| /external/grub/netboot/ |
| sis900.c | 563 /* disable packet filtering before setting filter */
579 /* enable packet filitering */
902 * Description: transmits a packet and waits for completion or timeout.
906 * unsigned short s: size of the data-part of the packet.
907 * char *p: the data for the packet.
917 const char *p) /* Packet */
943 /* pad to minimum packet size */
955 printf("sis900_transmit: Queued Tx packet size %d.\n", (int) s);
967 /* packet unsuccessfully transmited */
978 * Description: checks for a received packet and returns it if found
[all...] |
| /external/chromium_org/content/renderer/media/ |
| webrtc_local_audio_source_provider_unittest.cc | 62 // Consume the first packet in the resampler, which contains only zero.
63 // And the consumption of the data will trigger pulling the real packet from
76 // Prepare the second packet for featching.
|
| /external/chromium_org/media/cast/logging/ |
| logging_raw.cc | 80 // Is this packet belonging to a new frame?
83 // Create a new entry - start with base packet map.
93 // Is this a new packet?
|
| /external/chromium_org/media/ffmpeg/ |
| ffmpeg_common.h | 55 AVPacket* packet = static_cast<AVPacket*>(x); local
56 av_free_packet(packet);
57 delete packet;
|
| /external/chromium_org/net/quic/congestion_control/ |
| channel_estimator.cc | 9 // scenario, the arriving packet pair will tend to arrive at the max bandwidth
36 // Old packet. The sequence_number use the full 64 bits even though it's
41 DVLOG(1) << "Skip channel estimator due to lost packet(s)";
|
| pacing_sender.cc | 57 // The next packet should be sent as soon as the current packets has
111 DVLOG(1) << "Delaying packet: "
118 DVLOG(1) << "Sending packet now";
|
| /external/chromium_org/net/tools/quic/test_tools/ |
| quic_test_utils.h | 68 const QuicEncryptedPacket& packet));
82 const QuicEncryptedPacket& packet) {
83 return QuicConnection::ProcessUdpPacket(self_address, peer_address, packet);
|
| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| relayserver_unittest.cc | 129 talk_base::TestClient::Packet* packet = client->NextPacket(); local
130 if (packet) {
131 talk_base::ByteBuffer buf(packet->buf, packet->size);
134 delete packet;
140 talk_base::TestClient::Packet* packet = client->NextPacket(); local
141 if (packet) {
142 raw = std::string(packet->buf, packet->size)
[all...] |
| /external/chromium_org/third_party/opus/src/silk/ |
| control.h | 65 /* I: Number of samples per packet in milliseconds; 10/20/40/60 */
71 /* I: Uplink packet loss in percent (0-100) */
128 /* I: Number of samples per packet in milliseconds; 10/20/40/60 */
|
| /external/dropbear/ |
| filelist.txt | 23 packet.c Handles the basic packet encryption/decryption,
24 and switching to the appropriate packet handlers.
|
| /external/iproute2/man/man8/ |
| tc-drr.8 | 18 traffic. There is no default class \-\- if a packet cannot be classified,
26 non-empty. This list is used for dequeuing. A packet is dequeued from
27 the class at the head of the list if the packet size is smaller or equal
|
| /external/iproute2/tc/ |
| tc_stab.c | 35 " mtu : max packet size we create rate map for {2047}\n"
37 " mpu : minimum packet size used in rate computations\n"
38 " overhead : per-packet size overhead used in rate computations\n"
|
