Searched refs:rtp_header_ (Results 1 – 7 of 7) sorted by relevance
/external/webrtc/modules/audio_coding/test/ |
D | target_delay_unittest.cc | 38 rtp_header_.payloadType = pltype; in SetUp() 39 rtp_header_.timestamp = 0; in SetUp() 40 rtp_header_.ssrc = 0x12345678; in SetUp() 41 rtp_header_.markerBit = false; in SetUp() 42 rtp_header_.sequenceNumber = 0; in SetUp() 84 rtp_header_.timestamp += kFrameSizeSamples; in Push() 85 rtp_header_.sequenceNumber++; in Push() 86 ASSERT_EQ(0, receiver_.InsertPacket(rtp_header_, in Push() 137 RTPHeader rtp_header_; member in webrtc::TargetDelayTest
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/external/webrtc/modules/audio_coding/acm2/ |
D | acm_receiver_unittest.cc | 53 rtp_header_.sequenceNumber = 0; in SetUp() 54 rtp_header_.timestamp = 0; in SetUp() 55 rtp_header_.markerBit = false; in SetUp() 56 rtp_header_.ssrc = 0x12345678; // Arbitrary. in SetUp() 57 rtp_header_.numCSRCs = 0; in SetUp() 58 rtp_header_.payloadType = 0; in SetUp() 115 rtp_header_.payloadType = payload_type; in SendData() 116 rtp_header_.timestamp = timestamp; in SendData() 119 rtp_header_, in SendData() 125 rtp_header_.sequenceNumber++; in SendData() [all …]
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D | audio_coding_module_unittest.cc | 182 rtp_utility_->Populate(&rtp_header_); in SetUp() 216 acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); in InsertPacket() 217 rtp_utility_->Forward(&rtp_header_); in InsertPacket() 246 RTPHeader rtp_header_; member in webrtc::AudioCodingModuleTestOldApi 635 rtp_utility_->Forward(&rtp_header_); in InsertPacket() 640 last_payload_vec_.size(), rtp_header_)); in InsertPacket() 769 uint32_t input_timestamp = rtp_header_.timestamp; in CbReceiveImpl() 775 EXPECT_EQ(rtp_header_.timestamp + kPacketSizeSamples, input_timestamp); in CbReceiveImpl() 776 EXPECT_EQ(rtp_header_.timestamp, info.encoded_timestamp); in CbReceiveImpl() 777 EXPECT_EQ(rtp_header_.payloadType, info.payload_type); in CbReceiveImpl() [all …]
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/external/webrtc/modules/audio_coding/neteq/ |
D | neteq_network_stats_unittest.cc | 236 kPayloadType, frame_size_samples_, &rtp_header_); in RunTest() 241 kPayloadType, frame_size_samples_, &rtp_header_); in RunTest() 244 ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header_, payload)); in RunTest() 322 RTPHeader rtp_header_; member in webrtc::test::NetEqNetworkStatsTest
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D | neteq_stereo_unittest.cc | 122 &rtp_header_); in GetNewPackets() 172 rtp_header_, rtc::ArrayView<const uint8_t>( in RunTest() 223 RTPHeader rtp_header_; member in webrtc::NetEqStereoTest
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/external/webrtc/modules/audio_coding/neteq/tools/ |
D | neteq_quality_test.cc | 403 kPayloadType, in_size_samples_, &rtp_header_); in Transmit() 409 rtp_header_, in Transmit()
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D | neteq_quality_test.h | 168 RTPHeader rtp_header_; variable
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