| /external/webrtc/modules/rtp_rtcp/source/ |
| D | video_rtp_depacketizer_h264.cc | 64 const uint8_t* const payload_data = rtp_payload.cdata(); in ProcessStapAOrSingleNalu() local
77 const uint8_t* nalu_start = payload_data + kNalHeaderSize; in ProcessStapAOrSingleNalu()
79 uint8_t nal_type = payload_data[0] & kH264TypeMask; in ProcessStapAOrSingleNalu()
94 nal_type = payload_data[kStapAHeaderSize] & kH264TypeMask; in ProcessStapAOrSingleNalu()
115 nalu.type = payload_data[start_offset] & kH264TypeMask; in ProcessStapAOrSingleNalu()
129 output_buffer.AppendData(payload_data, start_offset); in ProcessStapAOrSingleNalu()
134 &payload_data[start_offset], end_offset - start_offset, &sps, in ProcessStapAOrSingleNalu()
160 &payload_data[end_offset], in ProcessStapAOrSingleNalu()
180 if (PpsParser::ParsePpsIds(&payload_data[start_offset], in ProcessStapAOrSingleNalu()
197 &payload_data[start_offset], end_offset - start_offset); in ProcessStapAOrSingleNalu()
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| D | video_rtp_depacketizer_generic.cc | 43 const uint8_t* payload_data = rtp_payload.cdata(); in Parse() local
45 uint8_t generic_header = payload_data[0]; in Parse()
64 .picture_id = ((payload_data[1] & 0x7F) << 8) | payload_data[2]; in Parse()
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| D | rtp_sender_audio.cc | 152 const uint8_t* payload_data, in SendAudio() argument
154 return SendAudio(frame_type, payload_type, rtp_timestamp, payload_data, in SendAudio()
163 const uint8_t* payload_data, in SendAudio() argument
256 if (payload_size == 0 || payload_data == NULL) { in SendAudio()
300 memcpy(payload, payload_data, payload_size); in SendAudio()
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| D | nack_rtx_unittest.cc | 161 for (size_t n = 0; n < sizeof(payload_data); n++) { in SetUp()
162 payload_data[n] = n % 10; in SetUp()
213 timestamp / 90, payload_data, video_header, 0)); in RunRtxTest()
235 uint8_t payload_data[65000]; member in webrtc::RtpRtcpRtxNackTest
263 timestamp / 90, payload_data, video_header, 0)); in TEST_F()
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| D | rtp_sender_audio.h | 51 const uint8_t* payload_data,
59 const uint8_t* payload_data,
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| /external/webrtc/audio/ |
| D | channel_send_frame_transformer_delegate.cc | 24 const uint8_t* payload_data, in TransformableOutgoingAudioFrame() argument
32 payload_(payload_data, payload_size), in TransformableOutgoingAudioFrame()
90 const uint8_t* payload_data, in Transform() argument
97 payload_data, payload_size, absolute_capture_timestamp_ms, ssrc)); in Transform()
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| D | channel_send_frame_transformer_delegate.h | 58 const uint8_t* payload_data,
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| /external/webrtc/modules/audio_coding/include/ |
| D | audio_coding_module.h | 44 const uint8_t* payload_data, in SendData() argument
49 return SendData(frame_type, payload_type, timestamp, payload_data, in SendData()
55 const uint8_t* payload_data, in SendData() argument
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| /external/libchrome/mojo/public/cpp/system/ |
| D | message_pipe.cc | 72 uint8_t* payload_data = reinterpret_cast<uint8_t*>(buffer); in ReadMessageRaw() local
74 std::copy(payload_data, payload_data + num_bytes, payload->begin()); in ReadMessageRaw()
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_send_test.cc | 128 const uint8_t* payload_data, in SendData() argument
135 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); in SendData()
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| D | acm_send_test.h | 59 const uint8_t* payload_data,
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| D | acm_receiver_unittest.cc | 112 const uint8_t* payload_data, in SendData() argument
123 rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes)); in SendData()
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| D | audio_coding_module_unittest.cc | 113 const uint8_t* payload_data, in SendData() argument
121 last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); in SendData()
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| /external/webrtc/audio/voip/ |
| D | audio_egress.cc | 113 const uint8_t* payload_data, in SendData() argument
117 rtc::ArrayView<const uint8_t> payload(payload_data, payload_size); in SendData()
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| D | audio_egress.h | 110 const uint8_t* payload_data,
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| /external/webrtc/modules/audio_coding/test/ |
| D | TestVADDTX.cc | 41 const uint8_t* payload_data, in SendData() argument
45 return next_->SendData(frame_type, payload_type, timestamp, payload_data, in SendData()
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| D | TestAllCodecs.h | 31 const uint8_t* payload_data,
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| D | TestStereo.h | 37 const uint8_t* payload_data,
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| D | TestVADDTX.h | 36 const uint8_t* payload_data,
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| D | TestAllCodecs.cc | 66 const uint8_t* payload_data, in SendData() argument
84 memcpy(payload_data_, payload_data, payload_size); in SendData()
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| D | TestStereo.cc | 46 const uint8_t* payload_data, in SendData() argument
64 receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header); in SendData()
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| /external/libnl/lib/netfilter/ |
| D | queue_msg.c | 245 const void *payload_data, unsigned payload_len) in nfnl_queue_msg_send_verdict_payload() argument
268 iov[2].iov_base = (void *) payload_data; in nfnl_queue_msg_send_verdict_payload()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | rtp_encode.cc | 111 const uint8_t* payload_data, in SendData() argument
148 fwrite(payload_data, sizeof(uint8_t), payload_len_bytes, out_file_), in SendData()
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| /external/webrtc/video/ |
| D | rtp_video_stream_receiver2_unittest.cc | 1202 rtc::CopyOnWriteBuffer payload_data({'1', '2', '3', '4'}); in TEST_P() local
1212 uint8_t* payload = packet_to_send.AllocatePayload(payload_data.size()); in TEST_P()
1213 memcpy(payload, payload_data.data(), payload_data.size()); in TEST_P()
1220 mock_on_complete_frame_callback_.AppendExpectedBitstream(payload_data.data(), in TEST_P()
1221 payload_data.size()); in TEST_P()
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| /external/cronet/base/ |
| D | enterprise_util_mac.mm | 83 NSString* payload_data =
88 [payload_data dataUsingEncoding:NSUTF8StringEncoding]
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