| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | fec_receiver_unittest.cc | 50 std::list<RtpPacket*>* media_rtp_packets, in GenerateFrame()
60 void VerifyReconstructedMediaPacket(const RtpPacket* packet, int times) { in VerifyReconstructedMediaPacket()
70 void BuildAndAddRedMediaPacket(RtpPacket* packet) { in BuildAndAddRedMediaPacket()
71 RtpPacket* red_packet = generator_->BuildMediaRedPacket(packet); in BuildAndAddRedMediaPacket()
79 RtpPacket* red_packet = generator_->BuildFecRedPacket(packet); in BuildAndAddRedFecPacket()
106 std::list<RtpPacket*> media_rtp_packets; in TEST_F()
113 std::list<RtpPacket*>::iterator it = media_rtp_packets.begin(); in TEST_F()
137 std::list<RtpPacket*> media_rtp_packets; in InjectGarbagePacketLength()
172 std::list<RtpPacket*> media_rtp_packets; in TEST_F()
180 std::list<RtpPacket*>::iterator it = media_rtp_packets.begin(); in TEST_F()
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| D | fec_test_helper.cc | 28 RtpPacket* FrameGenerator::NextPacket(int offset, size_t length) { in NextPacket()
29 RtpPacket* rtp_packet = new RtpPacket; in NextPacket()
47 RtpPacket* FrameGenerator::BuildMediaRedPacket(const RtpPacket* packet) { in BuildMediaRedPacket()
49 RtpPacket* red_packet = new RtpPacket; in BuildMediaRedPacket()
64 RtpPacket* FrameGenerator::BuildFecRedPacket(const Packet* packet) { in BuildFecRedPacket()
67 RtpPacket* red_packet = NextPacket(0, packet->length + 1); in BuildFecRedPacket()
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| D | fec_test_helper.h | 25 struct RtpPacket : public Packet { struct
37 RtpPacket* NextPacket(int offset, size_t length); argument
40 RtpPacket* BuildMediaRedPacket(const RtpPacket* packet);
45 RtpPacket* BuildFecRedPacket(const Packet* packet);
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| D | producer_fec_unittest.cc | 115 std::list<RtpPacket*> rtp_packets; in TEST_F()
120 RtpPacket* rtp_packet = generator_->NextPacket(i, 10); in TEST_F()
156 std::list<RtpPacket*> rtp_packets; in TEST_F()
162 RtpPacket* rtp_packet = generator_->NextPacket(i * kNumPackets + j, 10); in TEST_F()
189 RtpPacket* packet = generator_->NextPacket(0, 10); in TEST_F()
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| D | producer_fec.h | 21 struct RtpPacket;
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| D | rtp_sender_video.h | 31 struct RtpPacket;
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| D | producer_fec.cc | 35 struct RtpPacket { struct
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| D | rtp_sender_video.cc | 32 struct RtpPacket { struct
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| /external/webrtc/webrtc/test/ |
| D | rtp_file_reader_unittest.cc | 33 test::RtpPacket packet; in CountRtpPackets()
74 test::RtpPacket packet; in CountRtpPackets()
84 test::RtpPacket packet; in CountRtpPacketsPerSsrc()
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| D | rtp_file_reader.h | 21 struct RtpPacket { struct
45 virtual bool NextPacket(RtpPacket* packet) = 0; argument
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| D | rtp_file_writer_unittest.cc | 31 test::RtpPacket packet; in WriteRtpPackets()
49 test::RtpPacket packet; in VerifyFileContents()
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| D | rtp_file_reader.cc | 94 virtual bool NextPacket(RtpPacket* packet) { in NextPacket()
96 packet->length = RtpPacket::kMaxPacketBufferSize; in NextPacket()
174 bool NextPacket(RtpPacket* packet) override { in NextPacket()
176 packet->length = RtpPacket::kMaxPacketBufferSize; in NextPacket()
342 bool NextPacket(RtpPacket* packet) override { in NextPacket()
343 uint32_t length = RtpPacket::kMaxPacketBufferSize; in NextPacket()
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| D | rtp_file_writer.h | 29 virtual bool WritePacket(const RtpPacket* packet) = 0;
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| D | rtp_file_writer.cc | 41 bool WritePacket(const RtpPacket* packet) override { in WritePacket()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | rtc_event_log_source.cc | 35 const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) { in GetRtpPacket()
40 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in GetRtpPacket()
81 const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event); in NextPacket()
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| D | rtpcat.cc | 40 webrtc::test::RtpPacket packet; in main()
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| D | rtp_file_source.cc | 58 RtpPacket temp_packet; in NextPacket()
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| /external/webrtc/webrtc/call/ |
| D | rtc_event_log2rtp_dump.cc | 127 const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in main()
143 webrtc::test::RtpPacket packet; in main()
183 webrtc::test::RtpPacket packet; in main()
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| D | rtc_event_log.proto | 49 optional RtpPacket rtp_packet = 3;
74 message RtpPacket { message
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_unittest_helper.h | 49 struct RtpPacket { struct
64 typedef std::list<RtpPacket*> PacketList; argument
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| D | remote_bitrate_estimator_unittest_helper.cc | 64 RtpPacket* packet = new RtpPacket; in GenerateFrame()
251 testing::RtpStream::RtpPacket* packet = packets.front(); in GenerateAndProcessFrame()
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
| D | rtp_to_text.cc | 36 webrtc::test::RtpPacket packet; in main()
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| D | bwe_rtp_play.cc | 61 webrtc::test::RtpPacket packet; in main()
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| /external/webrtc/webrtc/video/ |
| D | replay.cc | 283 test::RtpPacket packet; in RtpReplay()
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| /external/webrtc/webrtc/modules/video_coding/test/ |
| D | rtp_player.cc | 452 test::RtpPacket next_packet_;
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