| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | packet.cc | 26 payload_(NULL), in Packet()
41 payload_(NULL), in Packet()
52 payload_(NULL), in Packet()
67 payload_(NULL), in Packet()
90 assert(payload_); in ExtractRedHeaders()
91 const uint8_t* payload_ptr = payload_; in ExtractRedHeaders()
140 payload_ = &payload_memory_[header_.headerLength]; in ParseHeader()
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| D | packet.h | 77 const uint8_t* payload() const { return payload_; } in payload()
107 const uint8_t* payload_; // First byte after header. variable
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| D | neteq_quality_test.cc | 252 payload_.reset(new uint8_t[max_payload_bytes_]); in NetEqQualityTest()
383 rtc::ArrayView<const uint8_t>(payload_.get(), payload_size_bytes_), in Transmit()
420 in_size_samples_, &payload_[0], in Simulate()
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| D | neteq_quality_test.h | 131 rtc::scoped_ptr<uint8_t[]> payload_; variable
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| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| D | audio_encoder_copy_red_unittest.cc | 83 MockEncodeHelper() : write_payload_(false), payload_(NULL) { in MockEncodeHelper()
94 memcpy(encoded, payload_, info_.encoded_bytes); in Encode()
101 uint8_t* payload_; member in webrtc::MockEncodeHelper
251 helper.payload_ = payload; in TEST_F()
266 helper.payload_[i] += 10; in TEST_F()
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| /external/webrtc/webrtc/modules/audio_coding/test/ |
| D | target_delay_unittest.cc | 50 WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_); in SetUp()
145 ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, in Push()
198 uint8_t payload_[kPayloadLenBytes]; member in webrtc::TargetDelayTest
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| /external/libchrome/base/ |
| D | pickle.h | 28 PickleIterator() : payload_(NULL), read_index_(0), end_index_(0) {} in PickleIterator()
100 const char* payload_; // Start of our pickle's payload. variable
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| D | pickle.cc | 24 : payload_(pickle.payload()), in PickleIterator()
56 const char* current_read_ptr = payload_ + read_index_; in GetReadPointerAndAdvance()
67 const char* current_read_ptr = payload_ + read_index_; in GetReadPointerAndAdvance()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_format_vp9_unittest.cc | 131 rtc::scoped_ptr<uint8_t[]> payload_; member in webrtc::RtpPacketizerVp9Test
138 payload_.reset(new uint8_t[payload_size]); in Init()
139 memset(payload_.get(), 7, payload_size); in Init()
143 packetizer_->SetPayloadData(payload_.get(), payload_size_, NULL); in Init()
154 EXPECT_EQ(packet[i], payload_[payload_pos_++]); in CheckPayload()
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| D | rtp_format_vp9.h | 90 const uint8_t* payload_; // The payload data to be packetized. variable
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| D | rtp_format_vp9.cc | 484 payload_(nullptr), in RtpPacketizerVp9()
517 payload_ = payload; in SetPayloadData()
611 &payload_[packet_info.payload_start_pos], packet_info.size); in WriteHeaderAndPayload()
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| D | rtp_sender_unittest.cc | 125 payload_(kPayload), in RtpSenderTest()
145 int payload_; member in webrtc::RtpSenderTest
156 EXPECT_EQ(payload_, rtp_header.payloadType); in VerifyRTPHeaderCommon()
213 EXPECT_EQ(payload_, rtp_header.payloadType); in VerifyCVOPacket()
1099 payload_ = kAudioPayload; in SetUp()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | neteq_network_stats_unittest.cc | 194 InsertPacket(rtp_header_, payload_, next_send_time); in RunTest()
270 uint8_t payload_[kPayloadSizeByte]; member in webrtc::test::NetEqNetworkStatsTest
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