| /external/webrtc/webrtc/test/testsupport/ |
| D | packet_reader.cc | 37 int PacketReader::NextPacket(uint8_t** packet_pointer) { in NextPacket() function in webrtc::test::PacketReader
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | constant_pcm_packet_source.cc | 38 Packet* ConstantPcmPacketSource::NextPacket() { in NextPacket() function in webrtc::test::ConstantPcmPacketSource
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| D | rtp_file_source.cc | 56 Packet* RtpFileSource::NextPacket() { in NextPacket() function in webrtc::test::RtpFileSource
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| D | rtc_event_log_source.cc | 78 Packet* RtcEventLogSource::NextPacket() { in NextPacket() function in webrtc::test::RtcEventLogSource
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | fec_test_helper.cc | 28 RtpPacket* FrameGenerator::NextPacket(int offset, size_t length) { in NextPacket() function in webrtc::FrameGenerator
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| D | rtp_format_video_generic.cc | 48 bool RtpPacketizerGeneric::NextPacket(uint8_t* buffer, in NextPacket() function in webrtc::RtpPacketizerGeneric
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| D | rtp_format_h264.cc | 252 bool RtpPacketizerH264::NextPacket(uint8_t* buffer, in NextPacket() function in webrtc::RtpPacketizerH264
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| D | rtp_format_vp8.cc | 200 bool RtpPacketizerVp8::NextPacket(uint8_t* buffer, in NextPacket() function in webrtc::RtpPacketizerVp8
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| D | rtp_format_vp9.cc | 548 bool RtpPacketizerVp9::NextPacket(uint8_t* buffer, in NextPacket() function in webrtc::RtpPacketizerVp9
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| D | acm_send_test_oldapi.cc | 76 Packet* AcmSendTestOldApi::NextPacket() { in NextPacket() function in webrtc::test::AcmSendTestOldApi
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| D | audio_coding_module_unittest_oldapi.cc | 1149 test::Packet* NextPacket() override { in NextPacket() function in webrtc::AcmSenderBitExactnessOldApi
1727 test::Packet* NextPacket() override { in NextPacket() function in webrtc::AcmSwitchingOutputFrequencyOldApi
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| /external/webrtc/webrtc/modules/video_coding/test/ |
| D | stream_generator.cc | 97 bool StreamGenerator::NextPacket(VCMPacket* packet) { in NextPacket() function in webrtc::StreamGenerator
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| D | rtp_player.cc | 349 virtual int NextPacket(int64_t time_now) { in NextPacket() function in webrtc::rtpplayer::RtpPlayerImpl
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| /external/webrtc/webrtc/test/ |
| D | rtp_file_reader.cc | 94 virtual bool NextPacket(RtpPacket* packet) { in NextPacket() function in webrtc::test::InterleavedRtpFileReader
174 bool NextPacket(RtpPacket* packet) override { in NextPacket() function in webrtc::test::RtpDumpReader
342 bool NextPacket(RtpPacket* packet) override { in NextPacket() function in webrtc::test::PcapReader
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| /external/webrtc/webrtc/base/ |
| D | testclient.cc | 54 TestClient::Packet* TestClient::NextPacket(int timeout_ms) { in NextPacket() function in rtc::TestClient
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | packet_buffer_unittest.cc | 52 Packet* PacketGenerator::NextPacket(int payload_size_bytes) { in NextPacket() function in webrtc::PacketGenerator
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