/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api.h | 58 const uint8_t* payload_data, 62 const uint8_t* payload_data() const { return payload_data_; } function in class:webrtc::TestRtpReceiver
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H A D | test_api.cc | 70 const uint8_t* payload_data, 74 memcpy(payload_data_, payload_data, payload_size); 69 OnReceivedPayloadData( const uint8_t* payload_data, const size_t payload_size, const webrtc::WebRtcRTPHeader* rtp_header) argument
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | acm_send_test_oldapi.cc | 112 const uint8_t* payload_data, 119 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); 108 SendData( FrameType frame_type, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, size_t payload_len_bytes, const RTPFragmentationHeader* fragmentation) argument
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_format_video_generic.cc | 33 const uint8_t* payload_data, 36 payload_data_ = payload_data; 91 const uint8_t* payload_data, 99 uint8_t generic_header = *payload_data++; 112 parsed_payload->payload = payload_data; 32 SetPayloadData( const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation) argument 90 Parse(ParsedPayload* parsed_payload, 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_format_vp8.cc | 184 const uint8_t* payload_data, 187 payload_data_ = payload_data; 668 const uint8_t* payload_data, 677 bool extension = (*payload_data & 0x80) ? true : false; // X bit 678 bool beginning_of_partition = (*payload_data & 0x10) ? true : false; // S bit 679 int partition_id = (*payload_data & 0x0F); // PartID field 688 (*payload_data & 0x20) ? true : false; // N bit 699 // Weak check for corrupt payload_data: PartID MUST NOT be larger than 8. 703 // Advance payload_data and decrease remaining payload size. 704 payload_data 183 SetPayloadData( const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation) argument [all...] |
H A D | rtp_receiver_audio.cc | 281 const uint8_t* payload_data, 312 bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false; 315 telephone_event_reported_.find(payload_data[4 * n]); 320 telephone_event_reported_.erase(payload_data[4 * n]); 326 telephone_event_reported_.insert(payload_data[4 * n]); 372 if (is_red && !(payload_data[0] & 0x80)) { 374 rtp_header->header.payloadType = payload_data[0]; 378 payload_data + 1, payload_length - 1, rtp_header); 383 payload_data, payload_length, rtp_header); 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 | nack_rtx_unittest.cc | 174 payload_data_length(sizeof(payload_data)), 216 payload_data[n] = n % 10; 266 timestamp / 90, payload_data, payload_data_length)); 289 uint8_t payload_data[65000]; member in class:webrtc::RtpRtcpRtxNackTest 311 timestamp / 90, payload_data, payload_data_length));
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H A D | rtp_format_h264.cc | 61 const uint8_t* payload_data, 70 const uint8_t* nalu_start = payload_data + kNalHeaderSize; 72 uint8_t nal_type = payload_data[0] & kTypeMask; 84 nal_type = payload_data[kStapAHeaderSize] & kTypeMask; 115 const uint8_t* payload_data, 122 uint8_t fnri = payload_data[0] & (kFBit | kNriMask); 123 uint8_t original_nal_type = payload_data[1] & kTypeMask; 124 bool first_fragment = (payload_data[1] & kSBit) > 0; 129 uint8_t* payload = const_cast<uint8_t*>(payload_data + *offset); 163 const uint8_t* payload_data, 60 ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, const uint8_t* payload_data, size_t payload_data_length) argument 114 ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload, const uint8_t* payload_data, size_t payload_data_length, size_t* offset) argument 162 SetPayloadData( const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation) argument 346 Parse(ParsedPayload* parsed_payload, const uint8_t* payload_data, size_t payload_data_length) argument [all...] |
H A D | rtp_sender_unittest.cc | 939 const uint8_t* payload_data = local 941 uint8_t generic_header = *payload_data++; 949 EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); 964 payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_); 965 generic_header = *payload_data++; 973 EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); 1222 const uint8_t* payload_data = local 1228 EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload))); 1251 const uint8_t* payload_data = local 1257 EXPECT_EQ(0, memcmp(payload, payload_data, sizeo [all...] |
H A D | rtp_rtcp_impl.cc | 405 const uint8_t* payload_data, 415 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, 400 SendOutgoingData( FrameType frame_type, int8_t payload_type, uint32_t time_stamp, int64_t capture_time_ms, const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_video_hdr) argument
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H A D | rtp_sender.cc | 505 const uint8_t* payload_data, 533 payload_data, payload_size, fragmentation); 544 capture_timestamp, capture_time_ms, payload_data, 501 SendOutgoingData(FrameType frame_type, int8_t payload_type, uint32_t capture_timestamp, int64_t capture_time_ms, const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_hdr) argument
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/external/webrtc/webrtc/video/ |
H A D | payload_router.cc | 55 const uint8_t* payload_data, 73 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, 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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H A D | vie_receiver.cc | 235 int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data, argument 241 if (vcm_->IncomingPacket(payload_data,
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/external/libnl/lib/netfilter/ |
H A D | queue_msg.c | 257 * @arg payload_data packet payload data 263 const void *payload_data, unsigned payload_len) 286 iov[2].iov_base = (void *) payload_data; 261 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/webrtc/webrtc/modules/audio_coding/test/ |
H A D | TestAllCodecs.cc | 61 uint32_t timestamp, const uint8_t* payload_data, 84 memcpy(payload_data_, payload_data, payload_size); 60 SendData(FrameType frame_type, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, size_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 size_t payload_size, const RTPFragmentationHeader* fragmentation) argument
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/external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
H A D | RTPencode.cc | 120 size_t makeDTMFpayload(unsigned char* payload_data, 1804 size_t makeDTMFpayload(unsigned char* payload_data, argument 1817 payload_data[0] = (unsigned char)Event; 1818 payload_data[1] = (unsigned char)(E | R | V); 1820 payload_data[2] = (unsigned char)((Duration >> 8) & 0xFF); 1821 payload_data[3] = (unsigned char)(Duration & 0xFF);
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