| /system/bt/vendor_libs/test_vendor_lib/include/ |
| packet.h | 51 // |header| and |payload| is copied into |header_| and |payload_|
71 std::vector<uint8_t> payload_; member in class:test_vendor_lib::Packet
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| /system/core/bootstat/ |
| event_log_list_builder.h | 50 // Returns true iff the remaining capacity in |payload_| is large enough to
59 // The size of the data stored in |payload_|. Used to track where to insert
65 std::unique_ptr<uint8_t[]> payload_; member in class:EventLogListBuilder
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| /system/connectivity/shill/net/ |
| netlink_packet.h | 84 ByteString* mutable_payload() { return payload_.get(); }
93 std::unique_ptr<ByteString> payload_; member in class:shill::NetlinkPacket
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| /system/update_engine/payload_generator/ |
| payload_file_unittest.cc | 35 PayloadFile payload_; member in class:chromeos_update_engine::PayloadFileTest
57 payload_.part_vec_.resize(2);
68 payload_.part_vec_[0].aops = aops;
72 payload_.part_vec_[1].aops = {aop};
74 EXPECT_TRUE(payload_.ReorderDataBlobs(orig_blobs, new_blobs));
76 const vector<AnnotatedOperation>& part0_aops = payload_.part_vec_[0].aops;
77 const vector<AnnotatedOperation>& part1_aops = payload_.part_vec_[1].aops;
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_format_vp9.h | 90 const uint8_t* payload_; // The payload data to be packetized. member in class:webrtc::RtpPacketizerVp9
|
| rtp_format_vp9_unittest.cc | 131 rtc::scoped_ptr<uint8_t[]> payload_; member in class:webrtc::RtpPacketizerVp9Test
138 payload_.reset(new uint8_t[payload_size]);
139 memset(payload_.get(), 7, payload_size);
143 packetizer_->SetPayloadData(payload_.get(), payload_size_, NULL);
154 EXPECT_EQ(packet[i], payload_[payload_pos_++]);
|
| rtp_sender_unittest.cc | 125 payload_(kPayload),
145 int payload_; member in class:webrtc::RtpSenderTest
156 EXPECT_EQ(payload_, rtp_header.payloadType);
213 EXPECT_EQ(payload_, rtp_header.payloadType);
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| packet.h | 77 const uint8_t* payload() const { return payload_; }
107 const uint8_t* payload_; // First byte after header. member in class:webrtc::test::Packet
|
| neteq_quality_test.h | 87 // |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
131 rtc::scoped_ptr<uint8_t[]> payload_; member in class:webrtc::test::NetEqQualityTest
|
| /external/libchrome/base/ |
| pickle.h | 28 PickleIterator() : payload_(NULL), read_index_(0), end_index_(0) {}
100 const char* payload_; // Start of our pickle's payload. member in class:base::PickleIterator
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| audio_encoder_copy_red_unittest.cc | 83 MockEncodeHelper() : write_payload_(false), payload_(NULL) {
94 memcpy(encoded, payload_, info_.encoded_bytes);
101 uint8_t* payload_; member in class:webrtc::MockEncodeHelper
251 helper.payload_ = payload;
266 helper.payload_[i] += 10;
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| neteq_network_stats_unittest.cc | 194 InsertPacket(rtp_header_, payload_, next_send_time);
270 uint8_t payload_[kPayloadSizeByte]; member in class:webrtc::test::NetEqNetworkStatsTest
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| target_delay_unittest.cc | 50 WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_);
145 ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2,
198 uint8_t payload_[kPayloadLenBytes]; member in class:webrtc::TargetDelayTest
|
