| /external/eigen/Eigen/src/Core/ |
| GenericPacketMath.h | 94 template<typename Packet> inline Packet
95 padd(const Packet& a,
96 const Packet& b) { return a+b; }
99 template<typename Packet> inline Packet
100 psub(const Packet& a,
101 const Packet& b) { return a-b; }
104 template<typename Packet> inline Packet
[all...] |
| Functors.h | 27 template<typename Packet>
28 EIGEN_STRONG_INLINE const Packet packetOp(const Packet& a, const Packet& b) const
30 template<typename Packet>
31 EIGEN_STRONG_INLINE const Scalar predux(const Packet& a) const
55 template<typename Packet>
56 EIGEN_STRONG_INLINE const Packet packetOp(const Packet& a, const Packet& b) cons
[all...] |
| /system/bt/vendor_libs/test_vendor_lib/src/ |
| packet.cc | 17 #define LOG_TAG "packet"
19 #include "vendor_libs/test_vendor_lib/include/packet.h"
31 Packet::Packet(serial_data_type_t type) : type_(type) {}
33 bool Packet::Encode(const std::vector<uint8_t>& header,
42 const std::vector<uint8_t>& Packet::GetHeader() const {
43 // Every packet must have a header.
48 uint8_t Packet::GetHeaderSize() const {
52 size_t Packet::GetPacketSize() const {
57 const std::vector<uint8_t>& Packet::GetPayload() const
[all...] |
| command_packet.cc | 29 CommandPacket::CommandPacket() : Packet(DATA_TYPE_COMMAND) {}
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| fec_test_helper.h | 23 typedef ForwardErrorCorrection::Packet Packet;
25 struct RtpPacket : public Packet {
39 // Creates a new RtpPacket with the RED header added to the packet.
40 RtpPacket* BuildMediaRedPacket(const RtpPacket* packet);
44 // header. Finally replaces the payload with the content of |packet->data|.
45 RtpPacket* BuildFecRedPacket(const Packet* packet);
47 void SetRedHeader(Packet* red_packet, uint8_t payload_type,
|
| forward_error_correction.h | 36 // TODO(holmer): As a next step all these struct-like packet classes should be
39 // and receiver_fec.cc to be refactored into the packet classes.
40 class Packet {
42 Packet() : length(0), data(), ref_count_(0) {}
43 virtual ~Packet() {}
52 size_t length; // Length of packet in bytes.
53 uint8_t data[IP_PACKET_SIZE]; // Packet data.
56 int32_t ref_count_; // Counts the number of references to a packet.
77 // media packet or higher than the last media packet.\
[all...] |
| /external/webrtc/webrtc/test/ |
| rtcp_packet_parser.cc | 25 const uint8_t* packet = static_cast<const uint8_t*>(data); local
26 RTCPUtility::RTCPParserV2 parser(packet, len, true);
32 sender_report_.Set(parser.Packet().SR);
35 receiver_report_.Set(parser.Packet().RR);
38 report_block_.Set(parser.Packet().ReportBlockItem);
39 ++report_blocks_per_ssrc_[parser.Packet().ReportBlockItem.SSRC];
45 sdes_chunk_.Set(parser.Packet().CName);
48 bye_.Set(parser.Packet().BYE);
51 app_.Set(parser.Packet().APP);
54 app_item_.Set(parser.Packet().APP)
[all...] |
| /external/apache-harmony/jdwp/src/test/java/org/apache/harmony/jpda/tests/framework/jdwp/ |
| ReplyPacket.java | 28 import org.apache.harmony.jpda.tests.framework.jdwp.Packet;
31 * This class represents JDWP reply packet.
33 public class ReplyPacket extends Packet {
52 * the JDWP packet, given as array of bytes.
57 Packet.SHORT_SIZE);
89 Packet.SHORT_SIZE);
|
| ParsedEvent.java | 44 protected ParsedEvent(byte suspendPolicy, Packet packet, byte eventKind) {
46 this.requestID = packet.getNextValueAsInt();
86 * @param packet
88 protected EventThread(byte suspendPolicy, Packet packet, byte eventKind) {
89 super(suspendPolicy, packet, eventKind);
90 this.threadID = packet.getNextValueAsThreadID();
112 * @param packet
114 protected EventThreadLocation(byte suspendPolicy, Packet packet
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| packet.h | 22 struct Packet {
31 Packet()
39 // Comparison operators. Establish a packet ordering based on (1) timestamp,
40 // (2) sequence number, (3) regular packet vs sync-packet and (4) redundancy.
43 // the packets is sync-packet, the regular packet is considered earlier. For
46 bool operator==(const Packet& rhs) const {
52 bool operator!=(const Packet& rhs) const { return !operator==(rhs); }
53 bool operator<(const Packet& rhs) const
[all...] |
| comfort_noise.h | 23 struct Packet;
48 // Update the comfort noise generator with the parameters in |packet|.
49 // Will delete the packet.
50 int UpdateParameters(Packet* packet);
|
| payload_splitter.h | 15 #include "webrtc/modules/audio_coding/neteq/packet.h"
43 // Splits each packet in |packet_list| into its separate RED payloads. Each
44 // RED payload is packetized into a Packet. The original elements in
52 // FEC as new packet for redundant decoding. The decoder database is needed to
53 // get information about which payload type each packet contains.
59 // is accepted. Any packet with another payload type is discarded.
66 // payload type each packet contains.
71 // Splits the payload in |packet|. The payload is assumed to be from a
73 virtual void SplitBySamples(const Packet* packet,
[all...] |
| /external/webrtc/webrtc/base/ |
| testclient.h | 24 // Records the contents of a packet that was received.
25 struct Packet {
26 Packet(const SocketAddress& a, const char* b, size_t s);
27 Packet(const Packet& p);
28 virtual ~Packet();
60 // Returns the next packet received by the client or 0 if none is received
61 // within the specified timeout. The caller must delete the packet
63 Packet* NextPacket(int timeout_ms);
65 // Checks that the next packet has the given contents. Returns the remot
[all...] |
| testclient.cc | 17 // DESIGN: Each packet received is put it into a list of packets.
23 packets_ = new std::vector<Packet*>();
54 TestClient::Packet* TestClient::NextPacket(int timeout_ms) {
56 // at most timeout_ms. If, during the loop, a packet arrives, then we can
59 // Note that the case where no packet arrives is important. We often want to
60 // test that a packet does not arrive.
77 // Return the first packet placed in the queue.
78 Packet* packet = NULL; local
81 packet = packets_->front()
91 Packet* packet = NextPacket(kTimeoutMs); local
103 Packet* packet = NextPacket(kNoPacketTimeoutMs); local
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| packet.h | 29 class Packet {
31 // Creates a packet, with the packet payload (including header bytes) in
34 // when the Packet object is deleted. The |time_ms| is an extra time
35 // associated with this packet, typically used to denote arrival time.
37 Packet(uint8_t* packet_memory,
45 // |virtual_packet_length_bytes| tells what size the packet had on wire,
48 Packet(uint8_t* packet_memory,
58 Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms);
60 Packet(uint8_t* packet_memory
[all...] |
| packet.cc | 11 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
21 Packet::Packet(uint8_t* packet_memory,
35 Packet::Packet(uint8_t* packet_memory,
50 Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms)
62 Packet::Packet(uint8_t* packet_memory,
77 bool Packet::ExtractRedHeaders(std::list<RTPHeader*>* headers) const
[all...] |
| rtc_event_log_source.h | 31 class Packet;
44 // Returns a pointer to the next packet. Returns NULL if end of file was
46 Packet* NextPacket() override;
|
| packet_source.h | 17 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
29 // Returns a pointer to the next packet. Returns NULL if the source is
31 virtual Packet* NextPacket() = 0;
44 // If SSRC filtering discards all packet that do not match the SSRC.
|
| /system/bt/vendor_libs/test_vendor_lib/include/ |
| packet.h | 31 class Packet {
33 virtual ~Packet() = default;
35 // Returns the size in octets of the entire packet, which consists of the type
49 // Validates the packet by checking that the payload size in the header is
54 // packet object will assume ownership of the copied data for its entire
60 // Constructs an empty packet of type |type|. A call to Encode() shall be made
61 // to check and fill in the packet's data.
62 Packet(serial_data_type_t type);
65 // Underlying containers for storing the actual packet, broken down into the
66 // packet header and the packet payload. Data is copied into the vector
[all...] |
| command_packet.h | 23 #include "vendor_libs/test_vendor_lib/include/packet.h"
43 class CommandPacket : public Packet {
71 // Size in octets of a command packet header, which consists of a 2 octet
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| packet.h | 26 class Packet {
30 Packet();
31 Packet(int flow_id, int64_t send_time_us, size_t payload_size);
32 virtual ~Packet();
34 virtual bool operator<(const Packet& rhs) const;
41 virtual Packet::Type GetPacketType() const = 0;
57 int64_t creation_time_us_; // Time when the packet was created.
58 int64_t send_time_us_; // Time the packet left last processor touching it.
59 int64_t sender_timestamp_us_; // Time the packet left the Sender.
64 class MediaPacket : public Packet {
[all...] |
| /external/eigen/Eigen/src/SparseLU/ |
| SparseLU_gemm_kernel.h | 21 * - lda and ldc must be multiples of the respective packet size
30 typedef typename packet_traits<Scalar>::type Packet;
74 Packet b00, b10, b20, b30, b01, b11, b21, b31;
75 b00 = pset1<Packet>(Bc0[0]);
76 b10 = pset1<Packet>(Bc0[1]);
77 if(RK==4) b20 = pset1<Packet>(Bc0[2]);
78 if(RK==4) b30 = pset1<Packet>(Bc0[3]);
79 b01 = pset1<Packet>(Bc1[0]);
80 b11 = pset1<Packet>(Bc1[1]);
81 if(RK==4) b21 = pset1<Packet>(Bc1[2])
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_send_test_oldapi.h | 27 class Packet;
47 // Returns the next encoded packet. Returns NULL if the test duration was
48 // exceeded. Ownership of the packet is handed over to the caller.
50 Packet* NextPacket();
65 // Creates a Packet object from the last packet produced by ACM (and received
66 // through the SendData method as a callback). Ownership of the new Packet
68 Packet* CreatePacket();
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
| after_initialization_fixture.h | 45 StorePacket(Packet::Rtp, data, len);
50 StorePacket(Packet::Rtcp, data, len);
70 struct Packet {
73 Packet() : len(0) {}
74 Packet(Type type, const void* data, size_t len)
84 void StorePacket(Packet::Type type,
89 packet_queue_.push_back(Packet(type, data, len));
110 Packet p;
119 if (p.type == Packet::Rtp) {
133 case Packet::Rtp
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/mock/ |
| mock_payload_splitter.h | 31 void(const Packet* packet, size_t bytes_per_ms,
34 int(const Packet* packet, size_t bytes_per_frame,
|
