| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | constant_pcm_packet_source.cc | 38 std::unique_ptr<Packet> ConstantPcmPacketSource::NextPacket() { in NextPacket() function in webrtc::test::ConstantPcmPacketSource
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| D | rtp_file_source.cc | 54 std::unique_ptr<Packet> RtpFileSource::NextPacket() { in NextPacket() function in webrtc::test::RtpFileSource
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| D | rtc_event_log_source.cc | 80 std::unique_ptr<Packet> RtcEventLogSource::NextPacket() { in NextPacket() function in webrtc::test::RtcEventLogSource
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| /external/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| D | common_header.h | 39 const uint8_t* NextPacket() const { in NextPacket() function
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_format_video_generic.cc | 52 bool RtpPacketizerGeneric::NextPacket(RtpPacketToSend* packet) { in NextPacket() function in webrtc::RtpPacketizerGeneric
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| D | rtp_format_vp8.cc | 76 bool RtpPacketizerVp8::NextPacket(RtpPacketToSend* packet) { in NextPacket() function in webrtc::RtpPacketizerVp8
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| D | fec_test_helper.cc | 119 std::unique_ptr<AugmentedPacket> AugmentedPacketGenerator::NextPacket( in NextPacket() function in webrtc::test::fec::AugmentedPacketGenerator
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| D | rtp_format_h264.cc | 237 bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet) { in NextPacket() function in webrtc::RtpPacketizerH264
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| D | rtp_packetizer_av1.cc | 350 bool RtpPacketizerAv1::NextPacket(RtpPacketToSend* packet) { in NextPacket() function in webrtc::RtpPacketizerAv1
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| D | rtp_format_vp9.cc | 333 bool RtpPacketizerVp9::NextPacket(RtpPacketToSend* packet) { in NextPacket() function in webrtc::RtpPacketizerVp9
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_send_test.cc | 98 std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() { in NextPacket() function in webrtc::test::AcmSendTestOldApi
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| D | audio_coding_module_unittest.cc | 1191 std::unique_ptr<test::Packet> NextPacket() override { in NextPacket() function in webrtc::AcmSenderBitExactnessOldApi
1840 std::unique_ptr<test::Packet> NextPacket() override { in NextPacket() function in webrtc::AcmSwitchingOutputFrequencyOldApi
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| /external/webrtc/modules/video_coding/test/ |
| D | stream_generator.cc | 95 bool StreamGenerator::NextPacket(VCMPacket* packet) { in NextPacket() function in webrtc::StreamGenerator
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| /external/webrtc/test/ |
| D | rtp_file_reader.cc | 84 bool NextPacket(RtpPacket* packet) override { in NextPacket() function in webrtc::test::InterleavedRtpFileReader
159 bool NextPacket(RtpPacket* packet) override { in NextPacket() function in webrtc::test::RtpDumpReader
322 bool NextPacket(RtpPacket* packet) override { in NextPacket() function in webrtc::test::PcapReader
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| /external/webrtc/rtc_base/ |
| D | test_client.cc | 63 std::unique_ptr<TestClient::Packet> TestClient::NextPacket(int timeout_ms) { in NextPacket() function in rtc::TestClient
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| /external/webrtc/modules/video_coding/ |
| D | loss_notification_controller_unittest.cc | 59 Packet NextPacket() { in NextPacket() function in webrtc::__anon08249e510111::PacketStreamCreator
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| /external/webrtc/modules/audio_coding/neteq/ |
| D | packet_buffer_unittest.cc | 77 webrtc::Packet PacketGenerator::NextPacket( in NextPacket() function in __anonc3ff7c1f0111::PacketGenerator
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| /external/webrtc/video/ |
| D | rtp_video_stream_receiver_unittest.cc | 284 RtpPacketReceived NextPacket() { in TEST_F() function in webrtc::TEST_F::__anone3011cd70210
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| D | rtp_video_stream_receiver2_unittest.cc | 289 RtpPacketReceived NextPacket() { in TEST_F() function in webrtc::TEST_F::__anon34ffb4890210
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