/external/eigen/Eigen/src/Core/ |
Diagonal.h | 153 // triger a compile time error is someone try to call packet 154 template<int LoadMode> typename MatrixType::PacketReturnType packet(Index) const; 155 template<int LoadMode> typename MatrixType::PacketReturnType packet(Index,Index) const;
|
MapBase.h | 113 inline PacketScalar packet(Index rowId, Index colId) const function in class:Eigen::MapBase 120 inline PacketScalar packet(Index index) const function in class:Eigen::MapBase
|
Replicate.h | 105 inline PacketScalar packet(Index rowId, Index colId) const function in class:Eigen::Replicate 114 return m_matrix.template packet<LoadMode>(actual_row, actual_col);
|
/external/libopus/src/ |
opus_decoder.c | 266 /* If we haven't got any packet yet, all we can do is return zeros */ 405 packet, so the exact behaviour is not normative. */ 668 /* Update the state as the last step to avoid updating it on an invalid packet */ 933 int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) 938 count = packet[0]&0x3; 946 return packet[1]&0x3F; 949 int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, 953 int count = opus_packet_get_nb_frames(packet, len); 958 samples = count*opus_packet_get_samples_per_frame(packet, Fs); 967 const unsigned char packet[], opus_int32 len [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
webrtcvideoengine2.cc | [all...] |
/external/chromium_org/media/cast/net/rtcp/ |
rtcp_builder_unittest.cc | 30 report_block.media_ssrc = kMediaSsrc; // SSRC of the RTP packet sender. 48 void ExpectPacketEQ(scoped_ptr<Packet> golden_packet, 49 PacketRef packet) { 50 EXPECT_EQ(golden_packet->size(), packet->data.size()); 51 if (golden_packet->size() == packet->data.size()) { 53 EXPECT_EQ((*golden_packet)[x], packet->data[x]); 54 if ((*golden_packet)[x] != packet->data[x])
|
/external/dnsmasq/src/ |
lease.c | 73 borrow DNS packet buffer which is always larger than 1000 bytes */ 77 daemon->dhcp_buff, daemon->packet) == 5) 88 if (strcmp(daemon->packet, "*") != 0) 89 clid_len = parse_hex(daemon->packet, (unsigned char *)daemon->packet, 255, NULL, NULL); 106 lease_set_hwaddr(lease, (unsigned char *)daemon->dhcp_buff2, (unsigned char *)daemon->packet, hw_len, hw_type, clid_len);
|
/external/iputils/ |
traceroute6.c | 65 * Attempt to trace the route an ip packet would follow to some 88 * traceroute to nis.nsf.net (35.1.1.48), 30 hops max, 56 byte packet 291 u_char packet[512]; /* last inbound (icmp) packet */ variable 479 "traceroute: packet size must be %d <= s < %d.\n", 611 if ((i = packet_ok(packet, cc, &from, &to, seq, &t1))) { 614 print(packet, cc, &from); 676 * "reset_timer" will only be true if the last packet that 690 iov.iov_base = packet; 691 iov.iov_len = sizeof(packet); [all...] |
/external/libvorbis/doc/ |
08-residue.tex | 16 residue vectors into the bitstream packet, and then reconstructs the 231 An end-of-packet condition at any point in header decode renders the 241 \subsubsection{packet decode} 243 Format 0 and 1 packet decode is identical except for specific 244 partition interleave. Format 2 packet decode can be built out of the 286 Packet decode proceeds as follows, matching the description offered earlier in the document. 302 9) [temp] = read from packet using codebook [residue_classbook] in scalar context 344 An end-of-packet condition during packet decode is to be considered a 367 3) vector [entry_temp] = read vector from packet using current codebook in VQ contex [all...] |
/external/openssl/ssl/ |
s23_clnt.c | 621 p=s->packet; 684 s->packet= &(s->s2->rbuf[0]); 685 memcpy(s->packet,buf,n); 786 s->packet= &(s->s3->rbuf.buf[0]); 787 memcpy(s->packet,buf,n);
|
d1_pkt.c | 196 s->packet = rdata->packet; 202 memcpy(&(s->s3->read_sequence[2]), &(rdata->packet[5]), 6); 229 rdata->packet = s->packet; 244 s->packet = NULL; 361 s->packet = rdata->packet; 392 * and we have that many bytes in s->packet 394 rr->input= &(s->packet[DTLS1_RT_HEADER_LENGTH]) [all...] |
/external/chromium_org/third_party/boringssl/src/ssl/ |
d1_pkt.c | 194 s->packet = rdata->packet; 200 memcpy(&(s->s3->read_sequence[2]), &(rdata->packet[5]), 6); 227 rdata->packet = s->packet; 234 s->packet = NULL; 351 s->packet = rdata->packet; 382 * and we have that many bytes in s->packet 384 rr->input= &(s->packet[DTLS1_RT_HEADER_LENGTH]) [all...] |
s23_clnt.c | 470 p=s->packet; 559 s->packet= &(s->s3->rbuf.buf[0]); 560 memcpy(s->packet,buf,n);
|
/device/asus/flo/camera/QCamera2/stack/mm-camera-interface/src/ |
mm_camera_stream.c | 1259 cam_sock_packet_t packet; local 1306 cam_sock_packet_t packet; local [all...] |
/device/lge/hammerhead/camera/QCamera2/stack/mm-camera-interface/src/ |
mm_camera_stream.c | 1251 cam_sock_packet_t packet; local 1298 cam_sock_packet_t packet; local [all...] |
/external/chromium_org/ui/events/gesture_detection/ |
touch_disposition_gesture_filter_unittest.cc | 119 const GestureEventDataPacket& packet) { 122 for (size_t i = 0; i < packet.gesture_count(); ++i) 123 touch_packet.Push(packet.gesture(i)); 134 SendGesturePacket(const GestureEventDataPacket& packet) { 135 return queue_->OnGesturePacket(packet); 455 // But other events in the same packet are still suppressed. 859 GestureEventDataPacket packet; local 861 SendGesturePacket(packet)); [all...] |
/external/libvorbis/lib/ |
info.c | 349 /* Is this packet a vorbis ID header? */ 355 oggpack_readinit(&opb,op->packet,op->bytes); 358 return(0); /* Not the initial packet */ 374 /* The Vorbis header is in three packets; the initial small packet in 375 the first page that identifies basic parameters, a second packet 376 with bitstream comments and a third packet that holds the 383 oggpack_readinit(&opb,op->packet,op->bytes); 399 /* Not the initial packet */ 555 op->packet = _ogg_malloc(oggpack_bytes(&opb)); 556 memcpy(op->packet, opb.buffer, oggpack_bytes(&opb)) [all...] |
/device/lge/mako/camera/QCamera/stack/mm-camera-interface/src/ |
mm_camera_channel.c | 1171 cam_sock_packet_t packet; local [all...] |
/external/deqp/modules/gles3/functional/ |
es3fShaderBuiltinVarTests.cpp | 910 rr::VertexPacket& packet = *packets[packetNdx]; local 913 packet.position = rr::readVertexAttribFloat(inputs[positionAttrLoc], packet.instanceNdx, packet.vertexNdx); 916 packet.outputs[VARYINGLOC_COLOR] = rr::readVertexAttribFloat(inputs[colorAttrLoc], packet.instanceNdx, packet.vertexNdx); 924 rr::FragmentPacket& packet = packets[packetNdx]; local 927 rr::writeFragmentOutput(context, packetNdx, fragNdx, 0, rr::readVarying<float>(packet, context, VARYINGLOC_COLOR, fragNdx)); [all...] |
es3fTextureUnitTests.cpp | 612 rr::VertexPacket& packet = *(packets[packetNdx]); local 614 packet.position = rr::readVertexAttribFloat(inputs[0], packet.instanceNdx, packet.vertexNdx); 615 packet.outputs[0] = rr::readVertexAttribFloat(inputs[1], packet.instanceNdx, packet.vertexNdx); 627 rr::FragmentPacket& packet = packets[packetNdx]; local 638 rr::readTriangleVarying<float>(packet, context, 0, 0).xy(), 639 rr::readTriangleVarying<float>(packet, context, 0, 1).xy() [all...] |
/external/chromium_org/remoting/host/ |
desktop_session_proxy.cc | 459 // Parse a serialized audio packet. No further validation is done since 461 scoped_ptr<AudioPacket> packet(new AudioPacket()); 462 if (!packet->ParseFromString(serialized_packet)) { 467 // Pass a captured audio packet to |audio_capturer_|. 470 base::Passed(&packet)));
|
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
datachannel.cc | 68 void DataChannel::PacketQueue::Push(DataBuffer* packet) { 69 byte_count_ += packet->size(); 70 packets_.push_back(packet);
|
/external/ipsec-tools/src/racoon/ |
isakmp_cfg.c | 135 * Handle an ISAKMP config mode packet 143 struct isakmp *packet; local 151 /* Check that the packet is long enough to have a header */ 152 if (msg->l < sizeof(*packet)) { 153 plog(LLV_ERROR, LOCATION, NULL, "Unexpected short packet\n"); 157 packet = (struct isakmp *)msg->v; 160 if ((packet->flags & ISAKMP_FLAG_E) == 0) { 167 * Decrypt the packet. If this is the beginning of a new 171 iph1->mode_cfg->last_msgid != packet->msgid ) 173 isakmp_cfg_newiv(iph1, packet->msgid) [all...] |
/external/chromium_org/media/audio/alsa/ |
alsa_output.cc | 232 // Buffer size is at least twice of packet size. 344 // If stopped, simulate a 0-length packet. 360 scoped_refptr<media::DataBuffer> packet = local 374 // Adjust packet size for downmix. 382 frames_filled, bytes_per_sample_, packet->writable_data()); 385 packet->set_data_size(packet_size); 386 // Add the packet to the buffer. 387 buffer_->Append(packet);
|
/external/chromium_org/media/cast/sender/ |
video_sender_unittest.cc | 66 // A singular packet implies a RTCP packet. 67 virtual bool SendPacket(PacketRef packet, 70 stored_packet_ = packet; 74 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { 77 // Check that at least one RTCP packet was sent before the first RTP 78 // packet. This confirms that the receiver will have the necessary lip 304 // Make sure that we send at least one RTCP packet. 311 // Build Cast msg and expect RTCP packet [all...] |