| /external/apache-harmony/jdwp/src/test/java/org/apache/harmony/jpda/tests/jdwp/ThreadReference/ |
| FramesTest.java | 429 CommandPacket packet = new CommandPacket( local
432 packet.setNextValueAsThreadID(threadID);
433 packet.setNextValueAsInt(startFrame);
434 packet.setNextValueAsInt(length);
435 ReplyPacket reply = debuggeeWrapper.vmMirror.performCommand(packet);
|
| /external/chromium_org/media/cast/net/ |
| cast_transport_sender_impl.cc | 241 NOTREACHED() << "Invalid request for sending RTCP packet.";
313 void CastTransportSenderImpl::OnReceivedPacket(scoped_ptr<Packet> packet) {
315 audio_rtcp_session_->IncomingRtcpPacket(&packet->front(),
316 packet->size())) {
320 video_rtcp_session_->IncomingRtcpPacket(&packet->front(),
321 packet->size())) {
324 VLOG(1) << "Stale packet received.";
|
| cast_transport_sender_impl_unittest.cc | 34 virtual bool SendPacket(PacketRef packet, const base::Closure& cb) OVERRIDE {
36 stored_packet_ = packet;
41 bytes_sent_ += packet->data.size();
199 // Resend packet 0.
223 // Resend one packet in the socket when unpaused.
224 // Resend one more packet from NACK.
262 // Resend one packet in the socket when unpaused.
299 // Resend one packet in the socket when unpaused.
365 // Retransmission of video packet now accepted.
|
| /external/chromium_org/remoting/codec/ |
| video_encoder_vpx.cc | 288 // TODO(hclam): Make sure we get exactly one frame from the packet.
290 scoped_ptr<VideoPacket> packet(
292 packet->mutable_format()->set_encoding(VideoPacketFormat::ENCODING_VP8);
303 packet->set_data(vpx_packet->data.frame.buf, vpx_packet->data.frame.sz);
311 packet->set_encode_time_ms(
314 return packet.Pass();
|
| /device/moto/shamu/camera/QCamera2/HAL/ |
| QCameraMem.cpp | 1113 struct encoder_media_buffer_type * packet = local
1159 struct encoder_media_buffer_type * packet = local
1184 struct encoder_media_buffer_type * packet = local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| mediachannel.h | 279 // Set DSCP value for packet sent from audio channel.
474 // Set DSCP value for packet sent from video channel.
555 rtc::Buffer* packet,
558 rtc::Buffer* packet,
574 // Called when a RTP packet is received.
575 virtual void OnPacketReceived(rtc::Buffer* packet,
577 // Called when a RTCP packet is received.
578 virtual void OnRtcpReceived(rtc::Buffer* packet,
614 // Base method to send packet using NetworkInterface.
615 bool SendPacket(rtc::Buffer* packet) {
[all...] |
| /external/chromium_org/third_party/webrtc/video/ |
| rampup_tests.cc | 106 // Just trigger if there was any rtx padding packet.
116 bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
119 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
140 packet,
153 bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
265 const uint8_t* packet, size_t length) {
268 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
279 bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
|
| /external/deqp/framework/referencerenderer/ |
| rrRenderer.cpp | 808 void transformVertexClipCoordsToWindowCoords (const RenderState& state, VertexPacket& packet)
812 packet.position = tcu::Vec4(packet.position.x()/packet.position.w(),
813 packet.position.y()/packet.position.w(),
814 packet.position.z()/packet.position.w(),
815 1.0f /packet.position.w());
828 packet.position = tcu::Vec4(packet.position.x()*halfW + oX
1048 const FragmentPacket& packet = fragmentPackets[packetNdx]; local
1080 const FragmentPacket& packet = fragmentPackets[packetNdx]; local
[all...] |
| /external/openssl/ssl/ |
| s2_pkt.c | 182 p=s->packet;
199 p=s->packet;
229 p= &(s->packet[2]);
282 /* Possibly the packet that we just read had 0 actual data bytes.
331 s->packet= &(s->s2->rbuf[s->s2->rbuf_offs]);
353 if (s->packet != s->s2->rbuf)
354 memcpy(s->s2->rbuf,s->packet,
372 s->packet=s->s2->rbuf;
|
| /external/deqp/modules/egl/ |
| teglRenderTests.cpp | 223 rr::VertexPacket& packet = *packets[packetNdx]; local
226 packet.position = rr::readVertexAttribFloat(inputs[positionAttrLoc], packet.instanceNdx, packet.vertexNdx);
229 packet.outputs[VaryingLoc_Color] = rr::readVertexAttribFloat(inputs[colorAttrLoc], packet.instanceNdx, packet.vertexNdx);
237 rr::FragmentPacket& packet = packets[packetNdx]; local
240 rr::writeFragmentOutput(context, packetNdx, fragNdx, 0, rr::readVarying<float>(packet, context, VaryingLoc_Color, fragNdx));
929 DrawOpPacket& packet = packets[threadNdx][packetNdx] local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_device/test/ |
| func_test_manager.cc | 157 AudioPacket* packet = new AudioPacket(); local
158 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample);
159 packet->nSamples = (uint16_t) nSamples;
160 packet->nBytesPerSample = nBytesPerSample;
161 packet->nChannels = nChannels;
162 packet->samplesPerSec = samplesPerSec;
163 _audioList.push_back(packet);
307 AudioPacket* packet = _audioList.front(); local
309 if (packet)
317 const uint16_t nSamplesIn = packet->nSamples
[all...] |
| /external/libvorbis/doc/ |
| 01-introduction.tex | 61 mechanism and decoder must allow that a packet may be any size, or
62 end before or after packet decode expects.
91 flag in each audio packet, or begin decode at any frame in the stream
96 Vorbis \emph{can} initiate decode at any arbitrary packet within a
168 low-level packet decode and synthesis.
267 specification. Once set up, decode may begin at any audio packet
294 \item decode packet type flag
317 \paragraph{Packet type decode}
319 Vorbis I uses four packet types. The first three packet types mark eac
[all...] |
| 04-codec.tex | 5 \section{Codec Setup and Packet Decode} \label{vorbis:spec:codec}
22 end-of-packet condition during decoding the first or third header
23 packet renders the stream undecodable. End-of-packet decoding the
28 Each header packet begins with the same header fields.
36 Decode continues according to packet type; the identification header
38 (these types are all odd as a packet with a leading single bit of '0'
39 is an audio packet). The packets must occur in the order of
277 \subsection{Audio packet decode and synthesis}
280 are audio. The first step of audio packet decode is to read an
[all...] |
| 05-comment.tex | 57 8) if ( [framing_bit] unset or end-of-packet ) then ERROR
200 header packet. Unlike the first bitstream header packet, it is not
201 generally the only packet on the second page and may not be restricted
203 packet is (practically) unbounded. The comment header packet is not
|
| /device/lge/mako/camera/ |
| QCameraHWI_Still.cpp | 2747 cam_sock_packet_t packet; local
2798 cam_sock_packet_t packet; local
2817 cam_sock_packet_t packet; local
[all...] |
| QCameraHWI_Record.cpp | 213 struct encoder_media_buffer_type * packet = local
216 native_handle_delete(const_cast<native_handle_t *>(packet->meta_handle));
461 struct encoder_media_buffer_type * packet = local
464 packet->meta_handle = native_handle_create(1, 2); //1 fd, 1 offset and 1 size
465 packet->buffer_type = kMetadataBufferTypeCameraSource;
466 native_handle_t * nh = const_cast<native_handle_t *>(packet->meta_handle);
|
| /device/lge/mako/camera/QCamera/HAL/core/src/ |
| QCameraHWI_Record.cpp | 214 struct encoder_media_buffer_type * packet = local
217 native_handle_delete(const_cast<native_handle_t *>(packet->meta_handle));
462 struct encoder_media_buffer_type * packet = local
465 packet->meta_handle = native_handle_create(1, 2); //1 fd, 1 offset and 1 size
466 packet->buffer_type = kMetadataBufferTypeCameraSource;
467 native_handle_t * nh = const_cast<native_handle_t *>(packet->meta_handle);
|
| /external/chromium_org/device/usb/ |
| usb_device_handle_impl.cc | 622 PlatformUsbIsoPacketDescriptor packet = &handle->iso_packet_desc[i]; local
623 if (packet->actual_length > 0) {
625 // all the data the packet can hold.
627 CHECK(packet_buffer_start + packet->actual_length <=
631 packet->actual_length);
633 actual_length += packet->actual_length;
636 packet_buffer_start += packet->length;
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| channel.h | 253 virtual bool SendPacket(rtc::Buffer* packet,
255 virtual bool SendRtcp(rtc::Buffer* packet,
270 bool SendPacket(bool rtcp, rtc::Buffer* packet,
272 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
273 void HandlePacket(bool rtcp, rtc::Buffer* packet,
695 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
|
| /external/chromium_org/net/quic/ |
| quic_dispatcher.cc | 78 // payload of the packet.
80 const QuicPublicResetPacket& /*packet*/) OVERRIDE {
84 const QuicVersionNegotiationPacket& /*packet*/) OVERRIDE {
209 const QuicEncryptedPacket& packet) {
212 current_packet_ = &packet;
213 // ProcessPacket will cause the packet to be dispatched in
216 framer_.ProcessPacket(packet);
217 // TODO(rjshade): Return a status describing if/why a packet was dropped,
235 // Ensure the packet has a version negotiation bit set before creating a new
237 // have the flag set. Otherwise it may be a stray packet
[all...] |
| /external/chromium_org/net/tools/quic/ |
| quic_dispatcher.cc | 83 // payload of the packet.
85 const QuicPublicResetPacket& /*packet*/) OVERRIDE {
89 const QuicVersionNegotiationPacket& /*packet*/) OVERRIDE {
214 const QuicEncryptedPacket& packet) {
217 current_packet_ = &packet;
218 // ProcessPacket will cause the packet to be dispatched in
221 framer_.ProcessPacket(packet);
222 // TODO(rjshade): Return a status describing if/why a packet was dropped,
240 // Ensure the packet has a version negotiation bit set before creating a new
242 // have the flag set. Otherwise it may be a stray packet
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_receiver.cc | 245 *RTCPArrivalTimeFrac = _lastReceivedSRNTPfrac; // local NTP time when we received a RTCP packet with a send block
326 // next top level packet.
404 const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
409 // The synchronization source identifier for the originator of this SR packet
412 // The source of the packet sender, same as of SR? or is this a CE?
483 // This will be called once per report block in the RTCP packet.
485 // Each packet has max 31 RR blocks.
729 // time since last received rtcp packet
737 // no rtcp packet for the last five regular intervals, reset limitations
804 const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
898 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local
908 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local
927 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| receiver.cc | 74 int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
77 // Insert the packet into the jitter buffer. The packet can either be empty or
80 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
93 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
267 // dual decoder has caught up with the decoder decoding with packet losses.
|
| /external/deqp/modules/gles2/functional/ |
| es2fTextureUnitTests.cpp | 456 rr::VertexPacket& packet = *(packets[packetNdx]); local
458 packet.position = rr::readVertexAttribFloat(inputs[0], packet.instanceNdx, packet.vertexNdx);
459 packet.outputs[0] = rr::readVertexAttribFloat(inputs[1], packet.instanceNdx, packet.vertexNdx);
471 rr::FragmentPacket& packet = packets[packetNdx]; local
482 rr::readTriangleVarying<float>(packet, context, 0, 0).xy(),
483 rr::readTriangleVarying<float>(packet, context, 0, 1).xy()
[all...] |
| /external/dnsmasq/src/ |
| util.c | 490 int read_write(int fd, unsigned char *packet, int size, int rw)
498 n = read(fd, &packet[done], (size_t)(size - done));
500 n = write(fd, &packet[done], (size_t)(size - done));
|
