/external/webrtc/api/ |
D | rtp_packet_info.cc | 27 int64_t receive_time_ms) in RtpPacketInfo() argument 33 receive_time_ms_(receive_time_ms) {} in RtpPacketInfo() 36 int64_t receive_time_ms) in RtpPacketInfo() argument 39 receive_time_ms_(receive_time_ms) { in RtpPacketInfo() 57 (lhs.receive_time_ms() == rhs.receive_time_ms()); in operator ==()
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D | rtp_packet_info.h | 38 int64_t receive_time_ms); 40 RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms); 67 int64_t receive_time_ms() const { return receive_time_ms_; } in receive_time_ms() function
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D | rtp_packet_info_unittest.cc | 162 EXPECT_EQ(rhs.receive_time_ms(), value); in TEST() 173 EXPECT_NE(rhs.receive_time_ms(), value); in TEST() 176 EXPECT_EQ(rhs.receive_time_ms(), value); in TEST()
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/external/webrtc/modules/video_coding/ |
D | packet.cc | 37 int64_t receive_time_ms) in VCMPacket() argument 50 packet_info(rtp_header, receive_time_ms) { in VCMPacket()
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D | packet.h | 44 int64_t receive_time_ms);
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D | frame_buffer2.cc | 231 int64_t receive_time_ms = first_frame->ReceivedTime(); in GetNextFrame() local 246 receive_time_ms = std::max(receive_time_ms, frame->ReceivedTime()); in GetNextFrame() 273 &frame_delay, receive_time_ms)) { in GetNextFrame()
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D | packet_buffer.h | 40 int64_t receive_time_ms);
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D | packet_buffer.cc | 42 int64_t receive_time_ms) in Packet() argument 55 receive_time_ms) {} in Packet()
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/external/webrtc/modules/remote_bitrate_estimator/ |
D | overuse_detector_unittest.cc | 98 int64_t receive_time_ms, in UpdateDetector() argument 104 rtp_timestamp, receive_time_ms, receive_time_ms, packet_size, in UpdateDetector() 108 overuse_detector_->State(), receive_time_ms); in UpdateDetector() 111 overuse_estimator_->num_of_deltas(), receive_time_ms); in UpdateDetector()
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/external/webrtc/modules/rtp_rtcp/source/ |
D | receive_statistics_impl.cc | 140 int64_t receive_time_ms) { in UpdateJitter() argument 141 int64_t receive_diff_ms = receive_time_ms - last_receive_time_ms_; in UpdateJitter()
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D | receive_statistics_impl.h | 53 void UpdateJitter(const RtpPacketReceived& packet, int64_t receive_time_ms)
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/external/webrtc/modules/audio_coding/neteq/ |
D | red_payload_splitter.cc | 127 /*receive_time_ms=*/red_packet.packet_info.receive_time_ms()); in SplitRed()
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D | neteq_impl_unittest.cc | 508 EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms); in TEST_F() 600 EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms); in TEST_F() 646 EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms); in TEST_F()
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D | neteq_impl.cc | 551 int64_t receive_time_ms = clock_->TimeInMilliseconds(); in InsertPacketInternal() local 556 packet_list.push_back([&rtp_header, &payload, &receive_time_ms] { in InsertPacketInternal() 563 packet.packet_info = RtpPacketInfo(rtp_header, receive_time_ms); in InsertPacketInternal()
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/external/webrtc/video/ |
D | rtp_video_stream_receiver2.cc | 735 min_recv_time = packet->packet_info.receive_time_ms(); in OnInsertedPacket() 736 max_recv_time = packet->packet_info.receive_time_ms(); in OnInsertedPacket() 742 std::min(min_recv_time, packet->packet_info.receive_time_ms()); in OnInsertedPacket() 744 std::max(max_recv_time, packet->packet_info.receive_time_ms()); in OnInsertedPacket()
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D | rtp_video_stream_receiver.cc | 764 min_recv_time = packet->packet_info.receive_time_ms(); in OnInsertedPacket() 765 max_recv_time = packet->packet_info.receive_time_ms(); in OnInsertedPacket() 771 std::min(min_recv_time, packet->packet_info.receive_time_ms()); in OnInsertedPacket() 773 std::max(max_recv_time, packet->packet_info.receive_time_ms()); in OnInsertedPacket()
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/external/webrtc/modules/congestion_controller/rtp/ |
D | transport_feedback_adapter_unittest.cc | 66 PacketResult CreatePacket(int64_t receive_time_ms, in CreatePacket() argument 72 res.receive_time = Timestamp::Millis(receive_time_ms); in CreatePacket()
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/external/webrtc/test/pc/e2e/analyzer/video/ |
D | default_video_quality_analyzer.cc | 358 return a.receive_time_ms() < b.receive_time_ms(); in OnFramePreDecode() 360 ->receive_time_ms(); in OnFramePreDecode()
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