/external/lldb/source/Utility/ |
H A D | 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;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | decode_bwe.c | 29 int32_t packet_size, 59 (int16_t) packet_size, /* in bytes */ 27 WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str, Bitstr_dec *streamdata, int32_t packet_size, uint16_t rtp_seq_number, uint32_t send_ts, uint32_t arr_ts) argument
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/external/lldb/test/pexpect-2.4/examples/ |
H A D | bd_client.py | 17 packet_size = cols * rows * 2 # double it for good measure 18 return s.recv(packet_size)
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | decode_bwe.c | 29 WebRtc_Word32 packet_size, 59 (WebRtc_Word16) packet_size, /* in bytes */ 27 WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str, Bitstr_dec *streamdata, WebRtc_Word32 packet_size, WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 send_ts, WebRtc_UWord32 arr_ts) argument
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/external/eigen/test/eigen2/ |
H A D | 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);
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/external/eigen/test/ |
H A D | 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);
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
H A D | codec_test.cc | 29 static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) { argument 31 if (packet_size == 160 || packet_size == 320) { 180 for (int packet_size = 80; packet_size < 1000; packet_size += 80) { 181 SetRateIfILBC(&codec_instance_, packet_size); 182 codec_instance_.pacsize = packet_size; 187 TEST_LOG("%d ", packet_size);
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | decode_bwe.c | 21 int32_t packet_size, 81 arrivalTimestampIn16kHz, packet_size); 18 WebRtcIsac_EstimateBandwidth( BwEstimatorstr* bwest_str, Bitstr* streamdata, int32_t packet_size, uint16_t rtp_seq_number, uint32_t send_ts, uint32_t arr_ts, enum IsacSamplingRate encoderSampRate, enum IsacSamplingRate decoderSampRate) argument
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | decode_bwe.c | 21 WebRtc_Word32 packet_size, 80 arrivalTimestampIn16kHz, packet_size); 18 WebRtcIsac_EstimateBandwidth( BwEstimatorstr* bwest_str, Bitstr* streamdata, WebRtc_Word32 packet_size, WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 send_ts, WebRtc_UWord32 arr_ts, enum IsacSamplingRate encoderSampRate, enum IsacSamplingRate decoderSampRate) argument
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/external/chromium_org/media/midi/ |
H A D | 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);
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/external/chromium_org/remoting/codec/ |
H A D | audio_encoder_opus_unittest.cc | 127 void TestEncodeDecode(int packet_size, argument 137 for (; pos < kTotalTestSamples; pos += packet_size) { 139 CreatePacket(packet_size, rate, frequency_hz, pos); 160 ValidateReceivedData(packet_size, kDefaultSamplingRate,
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/external/chromium_org/content/browser/renderer_host/p2p/ |
H A D | socket_host_tcp.cc | 484 int packet_size = base::NetToHost16(*reinterpret_cast<uint16*>(input)); local 485 if (input_len < packet_size + kPacketHeaderSize) 490 std::vector<char> data(cur, cur + packet_size); 492 consumed += packet_size; 533 int packet_size = GetExpectedPacketSize( local 536 if (input_len < packet_size + pad_bytes) 542 std::vector<char> data(cur, cur + packet_size); 544 consumed += packet_size; 596 int packet_size = base::NetToHost16(*reinterpret_cast<const uint16*>( local 605 packet_size [all...] |
/external/chromium_org/tools/usb_gadget/ |
H A D | hid_gadget.py | 28 packet_size=64, interval_ms=10, out_endpoint=True, 37 packet_size: Maximum interrupt packet size. 90 wMaxPacketSize=packet_size, 102 wMaxPacketSize=packet_size, 110 wMaxPacketSize=packet_size, 116 wMaxPacketSize=packet_size,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api_video.cc | 168 int packet_size = PaddingPacket(padding_packet, timestamp, seq_num, local 173 EXPECT_TRUE(parser->Parse(padding_packet, packet_size, &header)); 178 const int payload_length = packet_size - header.headerLength;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface/ |
H A D | isacfix.h | 194 * - packet_size : size of the packet. 205 int32_t packet_size, 217 * - packet_size : size of the packet. 230 int32_t packet_size,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/util/ |
H A D | 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) 136 get_arrival_time( int current_framesamples, int packet_size, int bottleneck, BottleNeckModel* BN_data, short senderSampFreqHz, short receiverSampFreqHz) argument
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H A D | utility.h | 102 int packet_size, /* bytes */
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/test/ |
H A D | stream_generator.cc | 48 const int packet_size = local 52 sequence_number_, timestamp_, packet_size, (i == 0), marker_bit, type));
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/interface/ |
H A D | 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,
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/external/chromium_org/third_party/mesa/src/src/mapi/glapi/gen/ |
H A D | 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()
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | TestAllCodecs.h | 64 int rate, int packet_size, int extra_byte);
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/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
H A D | overuse_detector.h | 28 void Update(uint16_t packet_size,
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/external/mesa3d/src/mapi/glapi/gen/ |
H A D | 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()
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/external/bluetooth/bluedroid/stack/rfcomm/ |
H A D | 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;
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
H A D | Isac_test.cc | 34 int packet_size, /* bytes */ 46 BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate); 33 get_arrival_time(int current_framesamples, int packet_size, int bottleneck, BottleNeckModel *BN_data) argument
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