/external/chromium_org/media/cast/net/rtp/ |
H A D | frame_buffer.cc | 25 bool FrameBuffer::InsertPacket(const uint8* payload_data, argument 55 payload_data, payload_data + payload_size, retval.first->second.begin());
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H A D | rtp_parser.cc | 24 const uint8** payload_data, 28 DCHECK(payload_data); 121 *payload_data = reinterpret_cast<const uint8*>(reader.ptr()); 21 ParsePacket(const uint8* packet, size_t length, RtpCastHeader* header, const uint8** payload_data, size_t* payload_size) argument
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H A D | framer.cc | 35 bool Framer::InsertPacket(const uint8* payload_data, argument 72 if (!it->second->InsertPacket(payload_data, payload_size, rtp_header)) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_send_test.cc | 96 const uint8_t* payload_data, 103 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); 93 SendData(FrameType frame_type, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, uint16_t payload_len_bytes, const RTPFragmentationHeader* fragmentation) argument
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H A D | acm_send_test_oldapi.cc | 100 const uint8_t* payload_data, 107 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); 96 SendData( FrameType frame_type, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, uint16_t payload_len_bytes, const RTPFragmentationHeader* fragmentation) argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_format_video_generic.cc | 32 const uint8_t* payload_data, 35 payload_data_ = payload_data; 94 const uint8_t* payload_data, 96 uint8_t generic_header = *payload_data++; 107 payload_data, payload_data_length, rtp_header) != 0) { 31 SetPayloadData( const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation) argument 93 Parse(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_data_length) argument
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H A D | rtp_format_vp8_test_helper.h | 41 uint8_t* payload_data() const { return payload_data_; } function in class:webrtc::test::RtpFormatVp8TestHelper
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H A D | rtp_receiver_audio.cc | 290 const uint8_t* payload_data, 321 bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false; 324 telephone_event_reported_.find(payload_data[4 * n]); 329 telephone_event_reported_.erase(payload_data[4 * n]); 335 telephone_event_reported_.insert(payload_data[4 * n]); 381 if (is_red && !(payload_data[0] & 0x80)) { 383 rtp_header->header.payloadType = payload_data[0]; 387 payload_data + 1, payload_length - 1, rtp_header); 392 payload_data, payload_length, rtp_header); 288 ParseAudioCodecSpecific( WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_length, const AudioPayload& audio_specific, bool is_red) argument
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H A D | rtp_format_h264.cc | 41 const uint8_t* payload_data, 49 uint8_t nal_type = payload_data[0] & kTypeMask; 51 nal_type = payload_data[3] & kTypeMask; 68 const uint8_t* payload_data, 71 uint8_t fnri = payload_data[0] & (kFBit | kNriMask); 72 uint8_t original_nal_type = payload_data[1] & kTypeMask; 73 bool first_fragment = (payload_data[1] & kSBit) > 0; 78 uint8_t* payload = const_cast<uint8_t*>(payload_data + *offset); 109 const uint8_t* payload_data, 114 payload_data_ = payload_data; 40 ParseSingleNalu(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_data_length) argument 67 ParseFuaNalu(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_data_length, size_t* offset) argument 108 SetPayloadData( const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation) argument 297 Parse(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_data_length) argument [all...] |
H A D | rtp_format_vp8.cc | 278 const uint8_t* payload_data, 281 payload_data_ = payload_data; 737 const uint8_t* payload_data, 740 if (!ParseVP8(rtp_header, payload_data, payload_data_length, &payload)) 277 SetPayloadData( const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation) argument
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H A D | nack_rtx_unittest.cc | 174 payload_data_length(sizeof(payload_data)), 218 payload_data[n] = n % 10; 270 payload_data, 294 uint8_t payload_data[65000]; member in class:RtpRtcpRtxNackTest 318 payload_data,
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H A D | rtp_sender_unittest.cc | 755 const uint8_t* payload_data = GetPayloadData(rtp_header, local 757 uint8_t generic_header = *payload_data++; 765 EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); 780 payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_); 781 generic_header = *payload_data++; 789 EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); 1029 const uint8_t* payload_data = GetPayloadData(rtp_header, local 1035 EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); 1058 const uint8_t* payload_data = GetPayloadData(rtp_header, local 1064 EXPECT_EQ(0, memcmp(payload, payload_data, sizeo [all...] |
H A D | rtp_rtcp_impl.cc | 508 const uint8_t* payload_data, 523 payload_data, 558 payload_data, 571 payload_data, 503 SendOutgoingData( FrameType frame_type, int8_t payload_type, uint32_t time_stamp, int64_t capture_time_ms, const uint8_t* payload_data, uint32_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_video_hdr) argument
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H A D | rtp_sender.cc | 407 const uint8_t *payload_data, const uint32_t payload_size, 433 payload_data, payload_size, fragmentation); 444 payload_data, payload_size, 404 SendOutgoingData( const FrameType frame_type, const int8_t payload_type, const uint32_t capture_timestamp, int64_t capture_time_ms, const uint8_t *payload_data, const uint32_t payload_size, const RTPFragmentationHeader *fragmentation, VideoCodecInformation *codec_info, const RTPVideoTypeHeader *rtp_type_hdr) argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api.h | 102 const uint8_t* payload_data() const { function in class:webrtc::TestRtpReceiver
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/external/libnl/lib/netfilter/ |
H A D | queue_msg.c | 220 * @arg payload_data packet payload data 226 const void *payload_data, unsigned payload_len) 249 iov[2].iov_base = (void *) payload_data; 224 nfnl_queue_msg_send_verdict_payload(struct nl_sock *nlh, const struct nfnl_queue_msg *msg, const void *payload_data, unsigned payload_len) argument
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/external/chromium_org/media/cast/receiver/ |
H A D | frame_receiver.cc | 80 const uint8* payload_data; local 85 &payload_data, 90 ProcessParsedPacket(rtp_header, payload_data, payload_size); 113 const uint8* payload_data, 128 framer_.InsertPacket(payload_data, payload_size, rtp_header, &duplicate); 112 ProcessParsedPacket(const RtpCastHeader& rtp_header, const uint8* payload_data, size_t payload_size) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | TestAllCodecs.cc | 56 uint32_t timestamp, const uint8_t* payload_data, 79 memcpy(payload_data_, payload_data, payload_size); 55 SendData(FrameType frame_type, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, uint16_t payload_size, const RTPFragmentationHeader* fragmentation) argument
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H A D | dual_stream_unittest.cc | 40 uint32_t timestamp, const uint8_t* payload_data, 298 const uint8_t* payload_data, 339 &payload_data[fragmentation->fragmentationOffset[n]], 384 payload_data, payload_size); 296 SendData(FrameType frameType, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, uint16_t payload_size, const RTPFragmentationHeader* fragmentation) argument
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H A D | TestStereo.cc | 50 const uint8_t* payload_data, 74 status = receiver_acm_->IncomingPacket(payload_data, payload_size, 47 SendData(const FrameType frame_type, const uint8_t payload_type, const uint32_t timestamp, const uint8_t* payload_data, const uint16_t payload_size, const RTPFragmentationHeader* fragmentation) argument
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/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_receiver.cc | 175 const uint8_t* payload_data, const uint16_t payload_size, 180 if (vcm_->IncomingPacket(payload_data, 174 OnReceivedPayloadData( const uint8_t* payload_data, const uint16_t payload_size, const WebRtcRTPHeader* rtp_header) argument
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H A D | vie_encoder.cc | 694 const uint8_t* payload_data, 703 payload_data, 689 SendData( const FrameType frame_type, const uint8_t payload_type, const uint32_t time_stamp, int64_t capture_time_ms, const uint8_t* payload_data, const uint32_t payload_size, const webrtc::RTPFragmentationHeader& fragmentation_header, const RTPVideoHeader* rtp_video_hdr) argument
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/external/chromium_org/net/spdy/ |
H A D | spdy_session_unittest.cc | 2732 char* payload_data = payload->data(); local 2733 test_stream.GetBytes(payload_data, kPayloadSize); 2736 framer.CreateDataFrame(1, payload_data, kPayloadSize, DATA_FLAG_NONE)); 2738 framer.CreateDataFrame(1, payload_data, kPayloadSize - 1, DATA_FLAG_FIN)); 2823 char* payload_data = payload->data(); local 2824 test_stream.GetBytes(payload_data, kPayloadSize); 2827 framer.CreateDataFrame(1, payload_data, kPayloadSize, DATA_FLAG_NONE));
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
H A D | RTPencode.cc | 82 int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration); 1743 int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration) { argument 1752 payload_data[0]=(unsigned char)Event; 1753 payload_data[1]=(unsigned char)(E|R|V); 1755 payload_data[2]=(unsigned char)((Duration>>8)&0xFF); 1756 payload_data[3]=(unsigned char)(Duration&0xFF);
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