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  /external/libogg/doc/libogg/
ogg_stream_packetpeek.html 20 <p>This function attempts to assemble a raw data packet and returns
26 the packet contents before handing it off to a codec for
28 the packet from the sync structure, call
52 <dd>Pointer to the next packet available in the bitstream, if
54 packet?" check.</dd>
61 <li>-1 if there's no packet available due to lost sync or a hole in the data.</li>
62 <li>0 if there is insufficient data available to complete a packet, or on unrecoverable internal error occurred.</li>
63 <li>1 if a packet is available.</li>
  /external/smack/src/org/jivesoftware/smackx/muc/
PacketMultiplexListener.java 28 import org.jivesoftware.smack.packet.Message;
29 import org.jivesoftware.smack.packet.Packet;
30 import org.jivesoftware.smack.packet.Presence;
44 public boolean accept(Packet packet) {
45 Message msg = (Message) packet;
80 public void processPacket(Packet p) {
  /prebuilts/gcc/linux-x86/host/i686-linux-glibc2.7-4.4.3/sysroot/usr/include/linux/
netfilter_bridge.h 15 /* If the packet is destined for this box. */
17 /* If the packet is destined for another interface. */
  /prebuilts/gcc/linux-x86/host/i686-linux-glibc2.7-4.6/sysroot/usr/include/linux/
netfilter_bridge.h 15 /* If the packet is destined for this box. */
17 /* If the packet is destined for another interface. */
  /prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.7-4.6/sysroot/usr/include/linux/
netfilter_bridge.h 15 /* If the packet is destined for this box. */
17 /* If the packet is destined for another interface. */
  /external/chromium_org/third_party/libjingle/source/talk/media/base/
rtpdump.cc 114 talk_base::StreamResult RtpDumpReader::ReadPacket(RtpDumpPacket* packet) {
115 if (!packet) return talk_base::SR_ERROR;
127 // Read the RTP dump packet header.
136 // Read the full length of the rtpdump packet, including the rtpdump header.
138 packet->data.resize(dump_packet_len - sizeof(header));
139 // Read the size of the original packet, which may be larger than the size in
140 // the rtpdump file, in the event that only part of the packet (perhaps just
144 packet->original_data_len = data_len;
145 // Read the elapsed time for this packet (different than RTP timestamp).
146 buf.ReadUInt32(&packet->elapsed_time)
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rtpdump.h 45 // For each packet, the file contains a 8 byte dump packet header, followed by
46 // the actual RTP or RTCP packet.
86 // packet. Return true and set the output parameter if successful.
92 // Get the type of the RTCP packet. Return true and set the output parameter
98 std::vector<uint8> data; // The actual RTP or RTCP packet.
99 size_t original_data_len; // The original length of the packet; may be
101 // packet was recorded.
117 virtual talk_base::StreamResult ReadPacket(RtpDumpPacket* packet);
146 virtual talk_base::StreamResult ReadPacket(RtpDumpPacket* packet);
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  /external/chromium_org/third_party/opus/src/tests/
test_opus_decode.c 62 unsigned char *packet; local
69 packet=malloc(sizeof(unsigned char)*MAX_PACKET);
70 if(packet==NULL)test_failed();
131 out_samples = opus_decode(dec[t], packet, 0, outbuf, 120/factor, fec);
136 out_samples = opus_decode(dec[t], packet, 0, outbuf, 0, fec);
138 out_samples = opus_decode(dec[t], packet, 0, 0, 0, fec);
143 out_samples = opus_decode(dec[t], packet, -1, outbuf, MAX_FRAME_SAMP, fec);
145 out_samples = opus_decode(dec[t], packet, INT_MIN, outbuf, MAX_FRAME_SAMP, fec);
147 out_samples = opus_decode(dec[t], packet, -1, outbuf, -1, fec);
151 out_samples = opus_decode(dec[t], packet, 1, outbuf, MAX_FRAME_SAMP, fec?-1:2)
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  /external/chromium_org/net/tools/quic/
quic_time_wait_list_manager.h 5 // Handles packets for guids in time wait state by discarding the packet and
6 // sending the clients a public reset packet with exponential backoff.
34 // Decides whether to send a public reset packet, a copy of the previously sent
35 // connection close packet, or nothing to the client which sent a packet
37 // period, a new connection/session will be created if a packet is received
50 // any packet bearing this guid should not be processed while the guid remains
52 // when packets are received for added guids. If NULL, a public reset packet
63 // Called when a packet is received for a guid that is in time wait state.
64 // Sends a public reset packet to the client which sent this guid. Sendin
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  /external/chromium_org/remoting/client/
audio_player_unittest.cc 94 scoped_ptr<AudioPacket> packet(new AudioPacket());
95 packet->set_encoding(AudioPacket::ENCODING_RAW);
96 packet->set_sampling_rate(rate);
97 packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
98 packet->set_channels(AudioPacket::CHANNELS_STEREO);
103 packet->add_data(data);
105 return packet.Pass();
121 scoped_ptr<AudioPacket> packet(CreatePacket44100Hz(10));
122 audio_->ProcessAudioPacket(packet.Pass());
144 // New packet with different sampling rate causes previous samples t
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rectangle_update_decoder.cc 42 virtual bool DecodePacket(const VideoPacket& packet) OVERRIDE {
43 return parent_->DecodePacket(packet);
118 void RectangleUpdateDecoder::DecodePacket(scoped_ptr<VideoPacket> packet,
127 // If the packet includes screen size or DPI information, store them.
128 if (packet->format().has_screen_width() &&
129 packet->format().has_screen_height()) {
130 webrtc::DesktopSize source_size(packet->format().screen_width(),
131 packet->format().screen_height());
138 if (packet->format().has_x_dpi() && packet->format().has_y_dpi())
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  /external/flac/libFLAC/
ogg_encoder_aspect.c 94 * packet. The packet is prefixed with
95 * + the one-byte packet type 0x7F
99 * - The first packet is flushed to the first page.
100 * - Each subsequent metadata block goes into its own packet.
101 * - Each metadata packet is flushed to page (this is not required,
104 * - Each subsequent FLAC audio frame goes into its own packet.
120 * Treat fLaC magic packet specially. We will note when we see it, then
121 * wait until we get the STREAMINFO and prepend it in that packet
124 ogg_packet packet; local
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  /external/speex/libspeex/
jitter.c 85 int curr_count; /**< Number of packet timings we got (including those we discarded) */
96 /* Add the timing of a new packet to the TimingBuffer */
100 /* Discard packet that won't make it into the list because they're too early */
140 spx_uint32_t last_returned_timestamp; /**< Useful for getting the next packet with the same timestamp (for fragmented media) */
146 spx_uint32_t arrival[SPEEX_JITTER_MAX_BUFFER_SIZE]; /**< Packet arrival time (0 means it was late, even though it's a valid timestamp) */
148 void (*destroy) (void *); /**< Callback for destroying a packet */
151 spx_int32_t concealment_size; /**< Size of the packet loss concealment "units" */
154 int late_cutoff; /**< How late must a packet be for it not to be considered at all */
192 /* Number of packet timings we have received (including those we didn't keep) */
199 /* Compute cost for one lost packet */
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  /external/smack/src/org/jivesoftware/smack/
Connection.java 43 import org.jivesoftware.smack.packet.Packet;
44 import org.jivesoftware.smack.packet.Presence;
146 * List of PacketListeners that will be notified when a new packet was received.
152 * List of PacketListeners that will be notified when a new packet was sent.
158 * List of PacketInterceptors that will be notified when a new packet is about to be
159 * sent to the server. These interceptors may modify the packet before it is being
415 * Sends the specified packet to the server.
417 * @param packet the packet to send
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Connection.java.orig 43 import org.jivesoftware.smack.packet.Packet;
44 import org.jivesoftware.smack.packet.Presence;
146 * List of PacketListeners that will be notified when a new packet was received.
152 * List of PacketListeners that will be notified when a new packet was sent.
158 * List of PacketInterceptors that will be notified when a new packet is about to be
159 * sent to the server. These interceptors may modify the packet before it is being
415 * Sends the specified packet to the server.
417 * @param packet the packet to send
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  /external/chromium_org/media/cast/rtp_receiver/rtp_parser/
rtp_parser.cc 27 bool RtpParser::ParsePacket(const uint8* packet, size_t length,
31 if (!ParseCommon(packet, length, rtp_header)) return false;
34 return ParseCast(packet + kRtpCommonHeaderLength,
41 bool RtpParser::ParseCommon(const uint8* packet,
45 uint8 version = packet[0] >> 6;
47 uint8 cc = packet[0] & 0x0f;
48 bool marker = ((packet[1] & 0x80) != 0);
49 int payload_type = packet[1] & 0x7f;
53 net::BigEndianReader big_endian_reader(packet + 2, 10);
76 bool RtpParser::ParseCast(const uint8* packet,
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  /external/chromium_org/remoting/codec/
video_decoder_verbatim.cc 29 bool VideoDecoderVerbatim::DecodePacket(const VideoPacket& packet) {
32 const char* in = packet.data().data();
34 for (int i = 0; i < packet.dirty_rects_size(); ++i) {
35 Rect proto_rect = packet.dirty_rects(i);
44 LOG(ERROR) << "Invalid packet received";
52 if (in + rect_row_size > packet.data().data() + packet.data().size()) {
53 LOG(ERROR) << "Invalid packet received";
62 if (in != packet.data().data() + packet.data().size())
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  /cts/tests/tests/net/src/android/net/ipv6/cts/
PingTest.java 53 byte[] packet = new byte[payloadLength + 8];
54 new Random().nextBytes(packet);
55 System.arraycopy(PING_HEADER, 0, packet, 0, PING_HEADER.length);
56 return packet;
78 * Sends a ping packet to a random port on the specified address on the specified socket.
81 InetAddress address, byte[] packet) throws ErrnoException, IOException {
85 // Send the packet.
86 int ret = Libcore.os.sendto(s, ByteBuffer.wrap(packet), 0, address, port);
87 assertEquals(packet.length, ret);
91 * Checks that a socket has received a response appropriate to the specified packet
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  /external/chromium_org/media/cast/net/rtp_sender/packet_storage/
packet_storage.cc 27 void Save(const Packet* packet) {
28 DCHECK_LT(packet->size(), kIpPacketSize) << "Invalid argument";
30 packet_.insert(packet_.begin(), packet->begin(), packet->end());
34 packets->push_back(Packet(packet_.begin(), packet_.end()));
38 Packet packet_;
68 // We should always find the packet.
74 // Add this packet to the free list for later re-use.
84 // Packet too old
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  /libcore/luni/src/main/java/java/net/
PlainDatagramSocketImpl.java 153 DatagramPacket packet = new DatagramPacket(EmptyArray.BYTE, 0); local
154 int result = peekData(packet);
156 sender.ipaddress = packet.getAddress().getAddress();
179 public void send(DatagramPacket packet) throws IOException {
180 int port = isNativeConnected ? 0 : packet.getPort();
181 InetAddress address = isNativeConnected ? null : packet.getAddress();
182 IoBridge.sendto(fd, packet.getData(), packet.getOffset(), packet.getLength(), 0, address, port);
228 * Set the received address and port in the packet. We do this when th
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  /external/chromium_org/net/quic/
quic_connection.cc 40 // If true, then QUIC connections will bundle acks with any outgoing packet when
42 // packet count of pure ack packets.
86 // This alarm will be scheduled any time a data-bearing packet is sent out.
105 // a delay before sending packets and fires when the packet may be sent.
227 delete it->packet;
274 const QuicPublicResetPacket& packet) {
276 debug_visitor_->OnPublicResetPacket(packet);
282 DVLOG(1) << ENDPOINT << "Received packet with mismatched version "
329 // TODO(satyamshekhar): Store the sequence number of this packet and close the
330 // connection if we ever received a packet with incorrect version and whos
1455 QuicEncryptedPacket* packet = undecryptable_packets_.front(); local
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  /external/smack/src/org/jivesoftware/smackx/
LastActivityManager.java 28 import org.jivesoftware.smack.packet.IQ;
29 import org.jivesoftware.smack.packet.Message;
30 import org.jivesoftware.smack.packet.Packet;
31 import org.jivesoftware.smack.packet.Presence;
32 import org.jivesoftware.smackx.packet.DiscoverInfo;
33 import org.jivesoftware.smackx.packet.LastActivity;
53 * the LastActivity packet to them, as in the following code:
69 * To get the uptime of a host, you simple send the LastActivity packet to it,
108 public void processPacket(Packet packet) {
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  /external/chromium_org/chrome/common/extensions/api/
diagnostics.idl 11 // Packet time to live value. If omitted, the system default value will be
14 // Packet timeout in seconds. If omitted, the system default value will be
23 // The IP may differs from our target IP if the packet's ttl is used up.
33 // Send a packet of the given type with the given parameters.
  /external/chromium_org/cloud_print/gcp20/prototype/
dns_packet_parser.cc 9 DnsPacketParser::DnsPacketParser(const char* packet, size_t length)
10 : packet_(packet),
12 record_parser_(packet, length, sizeof(header_)) {
13 net::BigEndianReader reader(packet, length);
  /external/chromium_org/net/quic/congestion_control/
inter_arrival_probe.h 25 // Call every time a packet is sent to the network.
28 // Call once for each sent packet that we receive an acknowledgement from
35 // Call once for each sent packet we receive a congestion feedback from the
37 // If a peer sends both and ack and feedback for a sent packet, both

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