/external/chromium_org/media/audio/alsa/ |
H A D | alsa_output.cc | 365 size_t packet_size = frames_filled * bytes_per_frame_; local 366 DCHECK_LE(packet_size, packet_size_); 375 packet_size = packet_size / bytes_per_frame_ * bytes_per_output_frame_; 384 if (packet_size > 0) { 385 packet->set_data_size(packet_size);
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/external/qemu/android/ |
H A D | shaper.c | 70 size_t packet_size = sizeof(*packet); local 73 packet_size += size; 75 packet = g_malloc(packet_size);
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | isacfix.c | 555 * - packet_size : size of the packet. 566 WebRtc_Word32 packet_size, 585 if (packet_size <= 0) { 589 } else if (packet_size > (STREAM_MAXW16<<1)) { 614 if (packet_size == 0) 623 packet_size, 648 * - packet_size : size of the packet. 660 WebRtc_Word32 packet_size, 680 if (packet_size <= 0) { 684 } else if (packet_size > (STREAM_MAXW1 564 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst, const WebRtc_UWord16 *encoded, WebRtc_Word32 packet_size, WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 arr_ts) argument 658 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst, const WebRtc_UWord16 *encoded, WebRtc_Word32 packet_size, WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 send_ts, WebRtc_UWord32 arr_ts) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | isacfix.c | 614 * - packet_size : size of the packet. 625 int32_t packet_size, 640 if (packet_size <= 0) { 644 } else if (packet_size > (STREAM_MAXW16<<1)) { 671 packet_size, 696 * - packet_size : size of the packet. 708 int32_t packet_size, 724 if (packet_size <= 0) { 728 } else if (packet_size > (STREAM_MAXW16<<1)) { 755 packet_size, 623 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst, const uint16_t *encoded, int32_t packet_size, uint16_t rtp_seq_number, uint32_t arr_ts) argument 706 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst, const uint16_t *encoded, int32_t packet_size, uint16_t rtp_seq_number, uint32_t send_ts, uint32_t arr_ts) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
H A D | kenny.cc | 53 int packet_size, /* bytes */ 65 BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate); 52 get_arrival_time(int current_framesamples, int packet_size, int bottleneck, BottleNeckModel *BN_data) argument
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/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
H A D | remote_bitrate_estimator_unittest_helper.cc | 59 int packet_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets); local 64 packet->size = packet_size;
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
H A D | kenny.c | 52 int packet_size, /* bytes */ 64 BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate); 51 get_arrival_time(int current_framesamples, int packet_size, int bottleneck, BottleNeckModel *BN_data) argument
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/external/bluetooth/bluedroid/stack/l2cap/ |
H A D | l2c_utils.c | 2643 UINT16 packet_size; local 2646 packet_size = btm_get_max_packet_size (p_ccb->p_lcb->remote_bd_addr); 2648 if (packet_size <= (L2CAP_PKT_OVERHEAD + L2CAP_FCR_OVERHEAD + L2CAP_SDU_LEN_OVERHEAD + L2CAP_FCS_LEN)) 2651 L2CAP_TRACE_ERROR ("l2cu_adjust_out_mps bad packet size: %u will use MPS: %u", packet_size, p_ccb->peer_cfg.fcr.mps); 2656 packet_size -= (L2CAP_PKT_OVERHEAD + L2CAP_FCR_OVERHEAD + L2CAP_SDU_LEN_OVERHEAD + L2CAP_FCS_LEN); 2666 if (p_ccb->peer_cfg.fcr.mps >= packet_size) 2667 p_ccb->tx_mps = p_ccb->peer_cfg.fcr.mps / packet_size * packet_size; 2671 L2CAP_TRACE_DEBUG ("l2cu_adjust_out_mps use %d Based on peer_cfg.fcr.mps: %u packet_size: %u", 2672 p_ccb->tx_mps, p_ccb->peer_cfg.fcr.mps, packet_size); [all...] |
/external/chromium_org/net/quic/ |
H A D | quic_connection_logger.cc | 39 size_t packet_size, 44 dict->SetInteger("size", packet_size); 52 size_t packet_size, 60 dict->SetInteger("size", packet_size); 37 NetLogQuicPacketCallback(const IPEndPoint* self_address, const IPEndPoint* peer_address, size_t packet_size, NetLog::LogLevel ) argument 48 NetLogQuicPacketSentCallback( QuicPacketSequenceNumber sequence_number, EncryptionLevel level, TransmissionType transmission_type, size_t packet_size, WriteResult result, NetLog::LogLevel ) argument
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H A D | quic_connection_test.cc | 823 QuicByteCount packet_size; local 825 .WillOnce(DoAll(SaveArg<3>(&packet_size), Return(true))); 833 return packet_size; 1225 QuicByteCount packet_size; local 1227 .WillOnce(DoAll(SaveArg<2>(&original), SaveArg<3>(&packet_size), 1240 OnPacketSent(_, _, _, packet_size - kQuicVersionSize, _)) 1943 QuicByteCount packet_size; local 1945 .WillOnce(DoAll(SaveArg<2>(&largest_observed), SaveArg<3>(&packet_size), 1958 OnPacketSent(_, _, _, packet_size - kQuicVersionSize, _));
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H A D | quic_framer.cc | 324 size_t packet_size) { 325 QuicDataWriter writer(packet_size); 427 DCHECK_LE(len, packet_size); 321 BuildDataPacket( const QuicPacketHeader& header, const QuicFrames& frames, size_t packet_size) argument
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/external/chromium_org/third_party/mesa/src/src/gallium/drivers/r300/ |
H A D | r300_emit.c | 833 unsigned packet_size = (vertex_array_count * 3 + 1) / 2; local 839 BEGIN_CS(2 + packet_size + vertex_array_count * 2); 840 OUT_CS_PKT3(R300_PACKET3_3D_LOAD_VBPNTR, packet_size);
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | isac.c | 991 * - packet_size : size of the packet. 1001 int32_t packet_size, 1020 if (packet_size < 10) { 1038 packet_size, rtp_seq_number, send_ts, 999 WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst, const uint16_t* encoded, int32_t packet_size, uint16_t rtp_seq_number, uint32_t send_ts, uint32_t arr_ts) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
H A D | RTPencode.cc | 245 int packet_size, fs; local 410 packet_size=atoi(argv[3]); 411 CHECK_NOT_NULL(packet_size); 412 printf("Packet size: %i\n",packet_size); 422 NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed); 528 NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD, numChannels); 561 len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels; 569 while (len==packet_size) { 622 enc_len=NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12] ,fs,&vad, useVAD, bitrate, numChannels); 749 packet_age += packet_size; [all...] |
/external/lldb/source/Plugins/Process/MacOSX-Kernel/ |
H A D | CommunicationKDP.cpp | 183 const size_t packet_size = request_packet.GetSize(); local 189 DumpPacket (log_strm, packet_data, packet_size); 195 packet_size, 199 if (bytes_written == packet_size) 203 log->Printf ("error: failed to send packet entire packet %" PRIu64 " of %" PRIu64 " bytes sent", (uint64_t)bytes_written, (uint64_t)packet_size);
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/external/mesa3d/src/gallium/drivers/r300/ |
H A D | r300_emit.c | 833 unsigned packet_size = (vertex_array_count * 3 + 1) / 2; local 839 BEGIN_CS(2 + packet_size + vertex_array_count * 2); 840 OUT_CS_PKT3(R300_PACKET3_3D_LOAD_VBPNTR, packet_size);
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | isac.c | 952 * - packet_size : size of the packet. 962 WebRtc_Word32 packet_size, 979 if (packet_size <= 0) { 997 packet_size, rtp_seq_number, send_ts, 960 WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst, const WebRtc_UWord16* encoded, WebRtc_Word32 packet_size, WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 send_ts, WebRtc_UWord32 arr_ts) argument
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/external/chromium_org/net/quic/test_tools/ |
H A D | quic_test_utils.cc | 67 size_t packet_size = GetPacketHeaderSize(header); local 69 DCHECK_LE(packet_size, max_plaintext_size); 73 frames[i], max_plaintext_size - packet_size, first_frame, last_frame, 77 packet_size += frame_size; 79 return framer->BuildDataPacket(header, frames, packet_size);
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/external/chromium_org/third_party/webrtc/base/ |
H A D | virtualsocketserver.cc | 812 size_t packet_size = data_size + UDP_HEADER_SIZE; local 813 if (socket->network_size_ + packet_size > network_capacity_) {
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