| /external/lldb/source/Utility/ |
| StringExtractorGDBRemote.cpp | 56 #define PACKET_MATCHES(s) ((packet_size == (sizeof(s)-1)) && (strcmp((packet_cstr),(s)) == 0))
57 #define PACKET_STARTS_WITH(s) ((packet_size >= (sizeof(s)-1)) && ::strncmp(packet_cstr, s, (sizeof(s)-1))==0)
63 const size_t packet_size = m_packet.size(); local
68 if (packet_size == 1) return eServerPacketType_interrupt;
72 if (packet_size == 1) return eServerPacketType_nack;
76 if (packet_size == 1) return eServerPacketType_ack;
112 if (packet_size == 2) return eServerPacketType_qC;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| decode_bwe.c | 29 int32_t packet_size,
59 (int16_t) packet_size, /* in bytes */
|
| /external/lldb/test/pexpect-2.4/examples/ |
| bd_client.py | 17 packet_size = cols * rows * 2 # double it for good measure
18 return s.recv(packet_size)
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| decode_bwe.c | 29 WebRtc_Word32 packet_size,
59 (WebRtc_Word16) packet_size, /* in bytes */
|
| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
| codec_test.cc | 29 static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) {
31 if (packet_size == 160 || packet_size == 320) {
153 for (int packet_size = 80; packet_size < 1000; packet_size += 80) {
154 SetRateIfILBC(&codec_instance_, packet_size);
155 codec_instance_.pacsize = packet_size;
160 TEST_LOG("%d ", packet_size);
|
| /external/eigen/test/eigen2/ |
| eigen2_first_aligned.cpp | 15 const int packet_size = sizeof(Scalar) * ei_packet_traits<Scalar>::size; local
16 VERIFY(((std::size_t(array) + sizeof(Scalar) * ei_alignmentOffset(array, size)) % packet_size) == 0);
|
| /external/eigen/test/ |
| first_aligned.cpp | 15 const int packet_size = sizeof(Scalar) * internal::packet_traits<Scalar>::size; local
16 VERIFY(((size_t(array) + sizeof(Scalar) * internal::first_aligned(array, size)) % packet_size) == 0);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| decode_bwe.c | 21 int32_t packet_size,
81 arrivalTimestampIn16kHz, packet_size);
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| decode_bwe.c | 21 WebRtc_Word32 packet_size,
80 arrivalTimestampIn16kHz, packet_size);
|
| /external/chromium_org/media/midi/ |
| usb_midi_input_stream.cc | 77 size_t packet_size = packet_size_table[code_index]; local
78 if (packet_size == 0) {
89 delegate_->OnReceivedData(it->second, &packet[1], packet_size, time);
|
| /external/chromium_org/remoting/codec/ |
| audio_encoder_opus_unittest.cc | 127 void TestEncodeDecode(int packet_size,
137 for (; pos < kTotalTestSamples; pos += packet_size) {
139 CreatePacket(packet_size, rate, frequency_hz, pos);
160 ValidateReceivedData(packet_size, kDefaultSamplingRate,
|
| /external/chromium_org/content/browser/renderer_host/p2p/ |
| socket_host_tcp.cc | 447 int packet_size = base::NetToHost16(*reinterpret_cast<uint16*>(input)); local
448 if (input_len < packet_size + kPacketHeaderSize)
453 std::vector<char> data(cur, cur + packet_size);
455 consumed += packet_size;
496 int packet_size = GetExpectedPacketSize( local
499 if (input_len < packet_size + pad_bytes)
505 std::vector<char> data(cur, cur + packet_size);
507 consumed += packet_size;
559 int packet_size = base::NetToHost16(*reinterpret_cast<const uint16*>( local
568 packet_size += kStunHeaderSize
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api_video.cc | 170 int packet_size = PaddingPacket(padding_packet, timestamp, seq_num, local
175 EXPECT_TRUE(parser->Parse(padding_packet, packet_size, &header));
180 const int payload_length = packet_size - header.headerLength;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface/ |
| isacfix.h | 195 * - packet_size : size of the packet.
206 int32_t packet_size,
218 * - packet_size : size of the packet.
231 int32_t packet_size,
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/util/ |
| utility.c | 138 int packet_size, /* bytes */
156 //travelTimeMs = ((packet_size + HeaderSize) * 8 * sampFreqHz) /
158 travelTimeMs = (unsigned int)floor((double)((packet_size + headerSizeByte) * 8 * 1000)
|
| utility.h | 102 int packet_size, /* bytes */
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/test/ |
| stream_generator.cc | 48 const int packet_size = local
52 sequence_number_, timestamp_, packet_size, (i == 0), marker_bit, type));
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/interface/ |
| isacfix.h | 195 * - packet_size : size of the packet.
206 WebRtc_Word32 packet_size,
218 * - packet_size : size of the packet.
231 WebRtc_Word32 packet_size,
|
| /external/chromium_org/media/audio/pulse/ |
| pulse_input.cc | 274 int packet_size = params_.GetBytesPerBuffer(); local
275 while (buffer_->forward_bytes() >= packet_size) {
276 buffer_->Read(audio_data_buffer_.get(), packet_size);
283 if (buffer_->forward_bytes() < packet_size)
|
| /external/chromium_org/third_party/mesa/src/src/mapi/glapi/gen/ |
| glX_doc.py | 63 def packet_size(self): member in class:glx_doc_parameter
107 [s, pad] = p.packet_size()
208 [s, pad] = output.packet_size()
223 [s, pad] = p.packet_size()
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| TestAllCodecs.h | 63 int rate, int packet_size, int extra_byte);
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| overuse_detector.h | 28 void Update(uint16_t packet_size,
|
| /external/mesa3d/src/mapi/glapi/gen/ |
| glX_doc.py | 63 def packet_size(self): member in class:glx_doc_parameter
107 [s, pad] = p.packet_size()
208 [s, pad] = output.packet_size()
223 [s, pad] = p.packet_size()
|
| /external/bluetooth/bluedroid/stack/rfcomm/ |
| port_utils.c | 145 UINT16 packet_size; local
151 packet_size = btm_get_max_packet_size (p_port->bd_addr);
152 if (packet_size == 0)
168 if ((L2CAP_MTU_SIZE + L2CAP_PKT_OVERHEAD) >= packet_size)
170 p_port->mtu = ((L2CAP_MTU_SIZE + L2CAP_PKT_OVERHEAD) / packet_size * packet_size) - RFCOMM_DATA_OVERHEAD - L2CAP_PKT_OVERHEAD;
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
| Isac_test.cc | 34 int packet_size, /* bytes */
46 BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
|
