/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | payload_splitter.cc | 61 new_packet->payload_length = red_packet->payload_length - sum_length; 71 new_packet->payload_length = ((payload_ptr[2] & 0x03) << 8) + 76 sum_length += new_packet->payload_length; 86 size_t payload_length = (*new_it)->payload_length; local 87 if (payload_ptr + payload_length > 88 red_packet->payload + red_packet->payload_length) { 102 (*new_it)->payload = new uint8_t[payload_length]; 103 memcpy((*new_it)->payload, payload_ptr, payload_length); [all...] |
H A D | packet.h | 25 size_t payload_length; member in struct:webrtc::Packet 33 payload_length(0),
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H A D | payload_splitter_unittest.cc | 47 void CreateOpusFecPayload(uint8_t* payload, size_t payload_length, argument 49 if (payload_length < 2) { 54 memset(&payload[2], payload_value, payload_length - 2); 83 packet->payload_length = (kPayloadLength + 1) + 85 uint8_t* payload = new uint8_t[packet->payload_length]; 123 Packet* CreatePacket(uint8_t payload_type, size_t payload_length, argument 129 packet->payload_length = payload_length; 130 uint8_t* payload = new uint8_t[packet->payload_length]; 133 CreateOpusFecPayload(packet->payload, packet->payload_length, 142 VerifyPacket(const Packet* packet, size_t payload_length, uint8_t payload_type, uint16_t sequence_number, uint32_t timestamp, uint8_t payload_value, bool primary = true) argument [all...] |
/external/wpa_supplicant_8/hostapd/src/wps/ |
H A D | ndef.c | 28 u32 payload_length; member in struct:ndef_record 46 record->payload_length = *pos++; 55 record->payload_length = len; 72 record->payload = record->payload_length == 0 ? NULL : pos; 73 pos += record->payload_length; 77 record->total_length < record->payload_length) 97 record.payload_length); 115 size_t payload_length = wpabuf_len(payload); local 117 short_record = payload_length < 256 ? 1 : 0; 124 total_len += type_length + id_length + payload_length; [all...] |
/external/wpa_supplicant_8/src/wps/ |
H A D | ndef.c | 28 u32 payload_length; member in struct:ndef_record 46 record->payload_length = *pos++; 55 record->payload_length = len; 72 record->payload = record->payload_length == 0 ? NULL : pos; 73 pos += record->payload_length; 77 record->total_length < record->payload_length) 97 record.payload_length); 115 size_t payload_length = wpabuf_len(payload); local 117 short_record = payload_length < 256 ? 1 : 0; 124 total_len += type_length + id_length + payload_length; [all...] |
/external/wpa_supplicant_8/wpa_supplicant/src/wps/ |
H A D | ndef.c | 28 u32 payload_length; member in struct:ndef_record 46 record->payload_length = *pos++; 55 record->payload_length = len; 72 record->payload = record->payload_length == 0 ? NULL : pos; 73 pos += record->payload_length; 77 record->total_length < record->payload_length) 97 record.payload_length); 115 size_t payload_length = wpabuf_len(payload); local 117 short_record = payload_length < 256 ? 1 : 0; 124 total_len += type_length + id_length + payload_length; [all...] |
/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_receiver_video.cc | 56 size_t payload_length, 64 RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength); 66 payload_length - rtp_header->header.paddingLength; 97 parsed_payload.payload_length, 52 ParseRtpPacket(WebRtcRTPHeader* rtp_header, const PayloadUnion& specific_payload, bool is_red, const uint8_t* payload, size_t payload_length, int64_t timestamp_ms, bool is_first_packet) argument
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H A D | producer_fec.h | 51 size_t payload_length, 56 size_t payload_length,
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H A D | rtp_receiver_audio.cc | 185 size_t payload_length, 202 payload_length, 282 size_t payload_length, 286 if (payload_length == 0) { 302 if (payload_length % 4 != 0) { 305 size_t number_of_events = payload_length / 4; 378 payload_data + 1, payload_length - 1, rtp_header); 383 payload_data, payload_length, rtp_header); 181 ParseRtpPacket(WebRtcRTPHeader* rtp_header, const PayloadUnion& specific_payload, bool is_red, const uint8_t* payload, size_t payload_length, int64_t timestamp_ms, bool is_first_packet) argument 279 ParseAudioCodecSpecific( WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_length, const AudioPayload& audio_specific, bool is_red) argument
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H A D | rtp_format.h | 57 size_t payload_length; member in struct:webrtc::RtpDepacketizer::ParsedPayload
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H A D | rtp_receiver_audio.h | 57 size_t payload_length, 99 size_t payload_length,
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H A D | producer_fec.cc | 124 size_t payload_length, 128 payload_length + kREDForFECHeaderLength + rtp_header_length); 132 red_packet->AssignPayload(data_buffer + rtp_header_length, payload_length); 137 size_t payload_length, 148 packet->length = payload_length + rtp_header_length; 123 BuildRedPacket(const uint8_t* data_buffer, size_t payload_length, size_t rtp_header_length, int red_pl_type) argument 136 AddRtpPacketAndGenerateFec(const uint8_t* data_buffer, size_t payload_length, size_t rtp_header_length) argument
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H A D | rtp_payload_registry_unittest.cc | 260 size_t payload_length, 263 new uint8_t[kRtxHeaderSize + header_length + payload_length](); 287 size_t payload_length = 200; local 288 size_t original_length = header_length + payload_length + kRtxHeaderSize; 300 header_length, payload_length, original_sequence_number)); 302 new uint8_t[header_length + payload_length]); 259 GenerateRtxPacket(size_t header_length, size_t payload_length, uint16_t original_sequence_number) argument
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H A D | rtp_sender_video.cc | 99 const size_t payload_length, 105 if (_rtpSender.SendToNetwork(data_buffer, payload_length, rtp_header_length, 108 _videoBitrate.Update(payload_length + rtp_header_length); 118 const size_t payload_length, 133 data_buffer, payload_length, rtp_header_length, red_payload_type_)); 135 producer_fec_.AddRtpPacketAndGenerateFec(data_buffer, payload_length, 98 SendVideoPacket(uint8_t* data_buffer, const size_t payload_length, const size_t rtp_header_length, uint16_t seq_num, const uint32_t capture_timestamp, int64_t capture_time_ms, StorageType storage) argument 117 SendVideoPacketAsRed(uint8_t* data_buffer, const size_t payload_length, const size_t rtp_header_length, uint16_t media_seq_num, const uint32_t capture_timestamp, int64_t capture_time_ms, StorageType media_packet_storage, bool protect) argument
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H A D | rtp_format_vp9.h | 104 size_t payload_length) override;
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H A D | rtp_receiver_strategy.h | 46 size_t payload_length,
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H A D | rtp_receiver_impl.cc | 163 size_t payload_length, 169 int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0; 174 if (payload_length == 0) { 187 size_t payload_data_length = payload_length - rtp_header.paddingLength; 202 &webrtc_rtp_header, payload_specific, is_red, payload, payload_length, 160 IncomingRtpPacket( const RTPHeader& rtp_header, const uint8_t* payload, size_t payload_length, PayloadUnion payload_specific, bool in_order) argument
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/external/tcpdump/ |
H A D | print-geonet.c | 111 uint16_t payload_length; local 128 payload_length = EXTRACT_16BITS(bp+4); 170 ND_PRINT((ndo, "Payload:%d ", payload_length));
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/external/webrtc/webrtc/test/ |
H A D | layer_filtering_transport.cc | 60 const size_t payload_length = length - header.headerLength; local 61 RTC_DCHECK_GT(payload_length, header.paddingLength); 62 const size_t payload_data_length = payload_length - header.paddingLength;
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H A D | rtp_file_reader.cc | 359 if (*length < next_packet_it_->payload_length) { 363 TRY_PCAP(Read(data, next_packet_it_->payload_length)); 364 *length = next_packet_it_->payload_length; 382 uint32_t payload_length; member in struct:webrtc::test::PcapReader::RtpPacketMarker 450 if (marker.payload_length > sizeof(read_buffer_)) { 454 TRY_PCAP(Read(read_buffer_, marker.payload_length)); 456 RtpUtility::RtpHeaderParser rtp_parser(read_buffer_, marker.payload_length); 570 marker->payload_length = length - kUdpHeaderLength;
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/external/v8/src/snapshot/ |
H A D | snapshot-common.cc | 176 int payload_length = startup_snapshot->RawData().length(); local 178 payload_length); 181 payload_length); 183 payload_offset += payload_length; 187 payload_length = context_snapshot->RawData().length(); 189 payload_length); 191 PrintF("%10d bytes for context #%d\n", payload_length, i); 193 payload_offset += payload_length;
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/external/webrtc/webrtc/video/ |
H A D | payload_router.cc | 56 size_t payload_length, 74 payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false; 51 RoutePayload(FrameType frame_type, int8_t payload_type, uint32_t time_stamp, int64_t capture_time_ms, const uint8_t* payload_data, size_t payload_length, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_video_hdr) argument
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
H A D | rtp_receiver.h | 75 size_t payload_length,
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | audio_coding_module_impl.h | 135 const size_t payload_length, 141 const size_t payload_length,
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/external/android-clat/ |
H A D | clatd_test.cpp | 203 size_t payload_length = 0; local 217 payload_length = len - sizeof(*ip); 218 pseudo_checksum = ipv4_pseudo_header_checksum(ip, payload_length); 237 payload_length = len - sizeof(*ip6) - sizeof(*frag); 243 payload_length = len - sizeof(*ip6); 247 if (payload_length) { 248 pseudo_checksum = ipv6_pseudo_header_checksum(ip6, payload_length, protocol); 258 if (payload_length) { 264 checksum = ip_checksum_finish(ip_checksum_add(pseudo_checksum, payload, payload_length)); 267 checksum = ip_checksum(payload, payload_length); 347 int payload_length = len - payload_offset; local [all...] |