/external/webrtc/modules/rtp_rtcp/source/ |
D | rtp_utility_unittest.cc | 24 const int8_t kPayloadType = 100; variable 34 0x80, kPayloadType, 0x00, kSeqNum, in TEST() 43 EXPECT_EQ(kPayloadType, header.payloadType); in TEST() 54 0x90, kPayloadType, 0x00, kSeqNum, in TEST() 67 EXPECT_EQ(kPayloadType, header.payloadType); in TEST() 80 0x90, kPayloadType, 0x00, kSeqNum, in TEST() 106 0x90, kPayloadType, 0x00, kSeqNum, in TEST() 131 0x90, kPayloadType, 0x00, kSeqNum, in TEST() 156 0x90, kPayloadType, 0x00, kSeqNum, in TEST() 227 0x90, kPayloadType, 0x00, kSeqNum, in TEST() [all …]
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D | rtp_packet_unittest.cc | 27 constexpr int8_t kPayloadType = 100; variable 48 0x80, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 53 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 60 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 68 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 79 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 89 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 98 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 107 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, 116 0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte, [all …]
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D | rtp_sender_audio_unittest.cc | 180 const uint8_t kPayloadType = 126; in TEST_F() local 182 kDtmfPayloadName, kPayloadType, kPayloadFrequency, 0, 0)); in TEST_F() 188 kPayloadName, kPayloadType, kPayloadFrequency, 1, 0)); in TEST_F() 197 AudioFrameType::kEmptyFrame, kPayloadType, capture_timestamp, nullptr, 0, in TEST_F() 204 kPayloadType, in TEST_F() 212 kPayloadType, in TEST_F()
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D | nack_rtx_unittest.cc | 38 const int kPayloadType = 123; variable 157 rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType); in SetUp() 210 kPayloadType, false)); in RunRtxTest() 213 kPayloadType, VideoCodecType::kVideoCodecGeneric, timestamp, in RunRtxTest() 233 {kRtxPayloadType, kPayloadType}}; 260 kPayloadType, false)); in TEST_F() 263 kPayloadType, VideoCodecType::kVideoCodecGeneric, timestamp, in TEST_F()
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D | rtp_sender_unittest.cc | 755 const uint8_t kPayloadType = 127; in TEST_P() local 771 kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp, in TEST_P() 783 kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp, in TEST_P() 796 kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp, in TEST_P() 810 kPayloadType, kCodecType, capture_time_ms * kCaptureTimeMsToRtpTimestamp, in TEST_P() 1247 const uint8_t kPayloadType = 127; in TEST_P() local 1260 ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, kCodecType, 1234, 4321, in TEST_P() 1276 ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, kCodecType, 1234, 4321, in TEST_P() 1288 const uint8_t kPayloadType = 111; in TEST_P() local 1301 ASSERT_TRUE(rtp_sender_video.SendVideo(kPayloadType, absl::nullopt, 1234, in TEST_P() [all …]
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/external/webrtc/modules/audio_coding/neteq/ |
D | decoder_database_unittest.cc | 40 const uint8_t kPayloadType = 0; in TEST() local 44 db.RegisterPayload(kPayloadType, SdpAudioFormat(kCodecName, 8000, 1))); in TEST() 47 EXPECT_EQ(DecoderDatabase::kOK, db.Remove(kPayloadType)); in TEST() 81 const uint8_t kPayloadType = 0; in TEST() local 85 db.RegisterPayload(kPayloadType, SdpAudioFormat(kCodecName, 8000, 1))); in TEST() 87 info = db.GetDecoderInfo(kPayloadType); in TEST() 91 EXPECT_EQ(decoder, db.GetDecoder(kPayloadType)); in TEST() 92 info = db.GetDecoderInfo(kPayloadType + 1); // Other payload type. in TEST() 98 const uint8_t kPayloadType = 0; in TEST() local 100 db.RegisterPayload(kPayloadType, SdpAudioFormat("l16", 8000, 1))); in TEST() [all …]
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D | neteq_impl_unittest.cc | 180 const uint8_t kPayloadType = 110; in TestDtmfPacket() local 188 rtp_header.payloadType = kPayloadType; in TestDtmfPacket() 194 kPayloadType, SdpAudioFormat("telephone-event", sample_rate_hz, 1))); in TestDtmfPacket() 289 const uint8_t kPayloadType = 0; in TEST_F() local 295 rtp_header.payloadType = kPayloadType; in TEST_F() 300 fake_packet.payload_type = kPayloadType; in TEST_F() 323 EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType)) in TEST_F() 332 .WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType), in TEST_F() 381 const uint8_t kPayloadType = 17; // Just an arbitrary number. in TEST_F() local 384 rtp_header.payloadType = kPayloadType; in TEST_F() [all …]
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D | neteq_decoder_plc_unittest.cc | 161 constexpr int kPayloadType = 100; in RunTest() local 164 encoder_config.payload_type = kPayloadType; in RunTest() 180 decoders.emplace(kPayloadType, SdpAudioFormat("l16", 32000, 1)); in RunTest()
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D | neteq_network_stats_unittest.cc | 137 static const uint8_t kPayloadType = 95; member in webrtc::test::NetEqNetworkStatsTest 174 neteq_->RegisterPayloadType(kPayloadType, format); in NetEqNetworkStatsTest() 236 kPayloadType, frame_size_samples_, &rtp_header_); in RunTest() 241 kPayloadType, frame_size_samples_, &rtp_header_); in RunTest()
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/external/webrtc/modules/audio_coding/neteq/tools/ |
D | neteq_performance_test.cc | 38 const int kPayloadType = 95; in Run() local 48 if (!neteq->RegisterPayloadType(kPayloadType, in Run() 69 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); in Run() 97 kPayloadType, kInputBlockSizeSamples, &rtp_header); in Run()
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D | packet_unittest.cc | 46 const uint8_t kPayloadType = 17; in TEST() local 50 MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc, in TEST() 56 EXPECT_EQ(kPayloadType, packet.header().payloadType); in TEST() 74 const uint8_t kPayloadType = 17; in TEST() local 78 MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc, in TEST() 85 EXPECT_EQ(kPayloadType, packet.header().payloadType); in TEST()
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D | neteq_quality_test.cc | 102 const uint8_t kPayloadType = 95; variable 307 ASSERT_TRUE(neteq_->RegisterPayloadType(kPayloadType, audio_format_)); in SetUp() 403 kPayloadType, in_size_samples_, &rtp_header_); in Transmit()
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/external/webrtc/video/ |
D | rtp_video_stream_receiver2_unittest.cc | 149 constexpr int kPayloadType = 100; variable 156 packet->SetPayloadType(kPayloadType); in CreateRtpPacketReceived() 183 codec.plType = kPayloadType; in RtpVideoStreamReceiver2Test() 374 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 392 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 423 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 445 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 502 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 539 rtp_packet.SetPayloadType(kPayloadType); in TEST_P() 581 constexpr int kPayloadType = 99; in TEST_P() local [all …]
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D | rtp_video_stream_receiver_unittest.cc | 148 constexpr int kPayloadType = 100; variable 155 packet->SetPayloadType(kPayloadType); in CreateRtpPacketReceived() 181 codec.plType = kPayloadType; in RtpVideoStreamReceiverTest() 369 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 387 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 418 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 440 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 496 rtp_packet.SetPayloadType(kPayloadType); in TEST_F() 533 rtp_packet.SetPayloadType(kPayloadType); in TEST_P() 575 constexpr int kPayloadType = 99; in TEST_P() local [all …]
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/external/webrtc/test/fuzzers/ |
D | audio_encoder_isac_fixed_fuzzer.cc | 20 constexpr int kPayloadType = 100; in FuzzOneInput() local 23 /*encoder=*/AudioEncoderIsacFix::MakeAudioEncoder(config, kPayloadType)); in FuzzOneInput()
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D | audio_encoder_isac_float_fuzzer.cc | 21 constexpr int kPayloadType = 100; in FuzzOneInput() local 24 config, kPayloadType)); in FuzzOneInput()
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D | audio_encoder_opus_fuzzer.cc | 21 constexpr int kPayloadType = 100; in FuzzOneInput() local 24 /*encoder=*/AudioEncoderOpus::MakeAudioEncoder(config, kPayloadType)); in FuzzOneInput()
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D | neteq_rtp_fuzzer.cc | 26 constexpr int kPayloadType = 95; variable 58 config.payload_type = kPayloadType; in FuzzRtpInput() 137 const auto it = codecs.find(kPayloadType); in FuzzOneInputTest()
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/external/webrtc/call/ |
D | rtp_video_sender_unittest.cc | 44 const int8_t kPayloadType = 96; variable 215 RtpVideoSenderTestFixture test({kSsrc1}, {kRtxSsrc1}, kPayloadType, {}); in TEST() 245 kPayloadType, {}); in TEST() 291 kPayloadType, {}); in TEST() 322 kPayloadType, {}); in TEST() 347 kPayloadType, states); in TEST() 370 RtpVideoSenderTestFixture test({kSsrc1}, {kRtxSsrc1}, kPayloadType, {}, in TEST() 419 kPayloadType, {}); in TEST() 584 kPayloadType, {}); in TEST() 681 RtpVideoSenderTestFixture test({kSsrc1}, {}, kPayloadType, {}); in TEST() [all …]
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/external/openscreen/cast/streaming/ |
D | rtp_packetizer_unittest.cc | 23 constexpr RtpPayloadType kPayloadType = RtpPayloadType::kAudioOpus; variable 87 EXPECT_EQ(kPayloadType, result->payload_type); in TestGeneratePacket() 131 RtpPacketizer packetizer_{kPayloadType, ssrc_,
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/external/webrtc/modules/audio_coding/acm2/ |
D | audio_coding_module_unittest.cc | 71 const uint8_t kPayloadType = 111; variable 167 : rtp_utility_(new RtpData(kFrameSizeSamples, kPayloadType)), in AudioCodingModuleTestOldApi() 203 acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}}); in RegisterCodec() 205 kPayloadType, *audio_format_, absl::nullopt)); in RegisterCodec() 351 acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}, in RegisterCngCodec() 621 acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}}); in RegisterCodec() 623 kPayloadType, *audio_format_, absl::nullopt)); in RegisterCodec() 708 config.payload_type = kPayloadType; in AcmReRegisterIsacMtTestOldApi() 729 acm_->SetReceiveCodecs({{kPayloadType, {"ISAC", kSampleRateHz, 1}}}); in RegisterCodec() 820 kPayloadType, *audio_format_, absl::nullopt)); in CbCodecRegistrationImpl() [all …]
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D | acm_receiver_unittest.cc | 411 const uint8_t kPayloadType = 111; in TEST_F() local 417 {{kPayloadType, SdpAudioFormat("L16", kSampleRateHz, 1)}}); in TEST_F() 420 rtp_header.payloadType = kPayloadType; in TEST_F()
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/external/webrtc/pc/ |
D | sdp_serializer.cc | 51 const char kPayloadType[] = "pt"; variable 284 builder << propertyDelimiter << kPayloadType << kDelimiterEqual; in SerializeRidDescription() 367 if (parts[0] == kPayloadType) { in DeserializeRidDescription()
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/external/webrtc/audio/voip/test/ |
D | audio_egress_unittest.cc | 245 constexpr int kPayloadType = 100; in TEST_F() local 250 egress_->RegisterTelephoneEventType(kPayloadType, kSampleRate); in TEST_F() 259 return (rtp.PayloadType() == kPayloadType && in TEST_F()
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/external/webrtc/modules/audio_coding/codecs/isac/ |
D | isac_webrtc_api_test.cc | 31 constexpr int kPayloadType = 42; variable 84 return AudioEncoderIsacFix::MakeAudioEncoder(config, kPayloadType); in CreateEncoder() 91 return AudioEncoderIsacFloat::MakeAudioEncoder(config, kPayloadType); in CreateEncoder()
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