| /external/chromium_org/media/midi/ |
| usb_midi_input_stream.h | 66 // Processes a USB-MIDI Event Packet.
67 // The first |kPacketSize| bytes of |packet| must be accessible.
70 const uint8* packet,
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| /external/chromium_org/remoting/client/plugin/ |
| media_source_video_renderer.h | 51 virtual void ProcessVideoPacket(scoped_ptr<VideoPacket> packet,
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| /external/chromium_org/remoting/codec/ |
| video_decoder_vpx.h | 29 virtual bool DecodePacket(const VideoPacket& packet) OVERRIDE;
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| video_decoder_vpx.cc | 95 bool VideoDecoderVpx::DecodePacket(const VideoPacket& packet) {
100 codec_.get(), reinterpret_cast<const uint8*>(packet.data().data()),
101 packet.data().size(), NULL, 0);
120 for (int i = 0; i < packet.dirty_rects_size(); ++i) {
121 Rect remoting_rect = packet.dirty_rects(i);
131 if (packet.has_use_desktop_shape()) {
132 for (int i = 0; i < packet.desktop_shape_rects_size(); ++i) {
133 Rect remoting_rect = packet.desktop_shape_rects(i);
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
| mock_packet_buffer.h | 31 int(Packet* packet));
44 Packet*(int* discard_count));
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| packet_buffer.h | 15 #include "webrtc/modules/audio_coding/neteq/packet.h"
48 // Inserts |packet| into the buffer. The buffer will take over ownership of
49 // the packet object.
52 virtual int InsertPacket(Packet* packet);
55 // ownership of the packet objects.
67 // Gets the timestamp for the first packet in the buffer and writes it to the
73 // Gets the timestamp for the first packet in the buffer with a timestamp no
81 // Returns a (constant) pointer the RTP header of the first packet in the
85 // Extracts the first packet in the buffer and returns a pointer to it
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_receiver_video.h | 32 const uint8_t* packet,
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| frame_buffer.h | 32 VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
53 // Get lowest packet sequence number in frame
55 // Get highest packet sequence number in frame
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| /external/chromium_org/third_party/webrtc/test/ |
| fake_network_pipe.h | 31 // TODO(mflodman) Add random and bursty packet loss.
50 // Random packet loss.
63 // Sends a new packet to the link.
64 void SendPacket(const uint8_t* packet, size_t packet_length);
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| /external/chromium_org/third_party/webrtc/video/ |
| rampup_tests.h | 55 virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE;
57 virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE;
106 virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
109 virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE;
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| /external/eigen/Eigen/src/Core/ |
| Swap.h | 34 typedef typename internal::packet_traits<Scalar>::type Packet;
99 Packet tmp = m_expression.template packet<StoreMode>(rowId, colId);
101 _other.template packet<LoadMode>(rowId, colId)
111 Packet tmp = m_expression.template packet<StoreMode>(index);
113 _other.template packet<LoadMode>(index)
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| CwiseBinaryOp.h | 179 EIGEN_STRONG_INLINE PacketScalar packet(Index rowId, Index colId) const function in class:Eigen::CwiseBinaryOpImpl
181 return derived().functor().packetOp(derived().lhs().template packet<LoadMode>(rowId, colId),
182 derived().rhs().template packet<LoadMode>(rowId, colId));
192 EIGEN_STRONG_INLINE PacketScalar packet(Index index) const function in class:Eigen::CwiseBinaryOpImpl
194 return derived().functor().packetOp(derived().lhs().template packet<LoadMode>(index),
195 derived().rhs().template packet<LoadMode>(index));
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| DiagonalProduct.h | 75 EIGEN_STRONG_INLINE PacketScalar packet(Index row, Index col) const function in class:Eigen::DiagonalProduct
87 EIGEN_STRONG_INLINE PacketScalar packet(Index idx) const function in class:Eigen::DiagonalProduct
92 return packet<LoadMode>(int(StorageOrder)==ColMajor?idx:0,int(StorageOrder)==ColMajor?0:idx);
99 return internal::pmul(m_matrix.template packet<LoadMode>(row, col),
110 return internal::pmul(m_matrix.template packet<LoadMode>(row, col),
111 m_diagonal.diagonal().template packet<DiagonalVectorPacketLoadMode>(id));
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| /external/nist-sip/java/gov/nist/javax/sip/stack/ |
| UDPMessageProcessor.java | 39 * packet, a new UDPMessageChannel is created (upto the max thread pool size).
190 DatagramPacket packet = new DatagramPacket(message, bufsize); local
191 sock.receive(packet);
240 this.messageQueue.add(packet);
244 new UDPMessageChannel(sipStack, this, packet);
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| /cts/hostsidetests/net/app/src/com/android/cts/net/hostside/ |
| VpnTest.java | 243 byte[] packet = new byte[LENGTH];
246 // Construct a ping packet.
248 random.nextBytes(packet);
257 System.arraycopy(header, 0, packet, 0, header.length);
259 // Send the packet.
262 Os.write(s, packet, 0, packet.length);
278 packet[4] = (byte) ((port >> 8) & 0xff);
279 packet[5] = (byte) (port & 0xff);
282 if (packet[0] == (byte) 0x80)
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
| vie_autotest_network.cc | 218 // Create a empty RTP packet.
219 unsigned char packet[3000]; local
220 memset(packet, 0, sizeof(packet));
221 packet[0] = 0x80; // V=2, P=0, X=0, CC=0
222 packet[1] = 0x7C; // M=0, PT = 124 (I420)
224 // Create a empty RTCP app packet.
233 tbChannel.videoChannel, packet, 1500));
238 tbChannel.videoChannel, packet, 1500));
242 tbChannel.videoChannel, packet, 11))
[all...] |
| /external/apache-harmony/jdwp/src/test/java/org/apache/harmony/jpda/tests/jdwp/MultiSession/ |
| ListenConnectorTest.java | 65 CommandPacket packet = new CommandPacket( local
69 ReplyPacket reply = debuggeeWrapper.vmMirror.performCommand(packet);
|
| /external/apache-harmony/jdwp/src/test/java/org/apache/harmony/jpda/tests/jdwp/VirtualMachine/ |
| CapabilitiesNewTest.java | 84 * there are no extra data in the reply packet;
89 CommandPacket packet = new CommandPacket( local
93 ReplyPacket reply = debuggeeWrapper.vmMirror.performCommand(packet);
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| CapabilitiesTest.java | 49 * there are no extra data in the reply packet;
54 CommandPacket packet = new CommandPacket( local
59 ReplyPacket reply = debuggeeWrapper.vmMirror.performCommand(packet);
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| ClassPathsTest.java | 54 * <BR> - there are no extra data in the reply packet;
59 CommandPacket packet = new CommandPacket( local
64 ReplyPacket reply = debuggeeWrapper.vmMirror.performCommand(packet);
|
| /external/chromium_org/media/cast/net/ |
| cast_transport_config.h | 124 typedef std::vector<uint8> Packet;
125 typedef scoped_refptr<base::RefCountedData<Packet> > PacketRef;
128 typedef base::Callback<void(scoped_ptr<Packet> packet)> PacketReceiverCallback;
132 // Send a packet to the network. Returns false if the network is blocked
137 virtual bool SendPacket(PacketRef packet, const base::Closure& cb) = 0;
161 uint32 media_ssrc; // SSRC of the RTP packet sender.
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| /external/chromium_org/media/filters/ |
| ffmpeg_video_decoder.cc | 260 // Create a packet for input data.
262 AVPacket packet; local
263 av_init_packet(&packet);
265 packet.data = NULL;
266 packet.size = 0;
268 packet.data = const_cast<uint8*>(buffer->data());
269 packet.size = buffer->data_size();
279 &packet);
286 // FFmpeg says some codecs might have multiple frames per packet. Previous
289 DCHECK_EQ(result, packet.size)
[all...] |
| /external/chromium_org/net/quic/ |
| quic_framer.h | 52 // Maximum number of missing packet ranges that can fit within an ack frame.
70 // Called only when |is_server_| is true and the the framer gets a packet with
71 // version flag true and the version on the packet doesn't match
74 // this packet.
77 // Called when a new packet has been received, before it
81 // Called when a public reset packet has been parsed but has not yet
84 const QuicPublicResetPacket& packet) = 0;
86 // Called only when |is_server_| is false and a version negotiation packet has
89 const QuicVersionNegotiationPacket& packet) = 0;
91 // Called when a lost packet has been recovered via FEC
[all...] |
| /device/moto/shamu/camera/QCamera/HAL/core/src/ |
| QCameraHWI_Record.cpp | 93 struct encoder_media_buffer_type * packet = local
95 native_handle_delete(const_cast<native_handle_t *>(packet->meta_handle));
221 struct encoder_media_buffer_type * packet = local
223 packet->meta_handle = native_handle_create(1, 2); //1 fd, 1 offset and 1 size
224 packet->buffer_type = kMetadataBufferTypeCameraSource;
225 native_handle_t * nh = const_cast<native_handle_t *>(packet->meta_handle);
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.cc | 62 RtpPacket* packet = new RtpPacket; local
63 packet->send_time = time_now_us + kSendSideOffsetUs;
64 packet->size = packet_size;
65 packet->rtp_timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
66 ((frequency_ / 1000) * packet->send_time + 500) / 1000);
67 packet->ssrc = ssrc_;
68 packets->push_back(packet);
79 // Generates an RTCP packet.
247 testing::RtpStream::RtpPacket* packet = packets.front(); local
249 // The simulated clock should match the time of packet->arrival_tim
[all...] |
