| /external/webrtc/modules/audio_coding/neteq/ |
| D | packet_buffer_unittest.cc | 50 webrtc::Packet NextPacket(
77 webrtc::Packet PacketGenerator::NextPacket( in NextPacket() function in __anona60d935a0111::PacketGenerator
122 const Packet packet = gen.NextPacket(payload_len, nullptr); in TEST()
148 buffer.InsertPacket(gen.NextPacket(payload_len, nullptr), &mock_stats)); in TEST()
172 buffer.InsertPacket(gen.NextPacket(payload_len, nullptr), &mock_stats)); in TEST()
179 const Packet packet = gen.NextPacket(payload_len, nullptr); in TEST()
202 list.push_back(gen.NextPacket(payload_len, nullptr)); in TEST()
241 list.push_back(gen.NextPacket(payload_len, nullptr)); in TEST()
245 Packet packet = gen.NextPacket(payload_len, nullptr); in TEST()
313 Packet packet = gen.NextPacket(kPayloadLength, nullptr); in TEST()
[all …]
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| /external/webrtc/test/fuzzers/ |
| D | rtp_packetizer_av1_fuzzer.cc | 49 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput()
55 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput()
60 RTC_CHECK(packetizer.NextPacket(&rtp_packet)) in FuzzOneInput()
66 RTC_CHECK(packetizer.NextPacket(&rtp_packet)); in FuzzOneInput()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_format_video_generic_unittest.cc | 39 while (packetizer->NextPacket(&packet)) { in NextPacketFillPayloadSizes()
84 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST()
138 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST()
154 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST()
166 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST()
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| D | rtp_format_vp9_unittest.cc | 159 EXPECT_TRUE(packetizer_->NextPacket(&packet_)); in CreateParseAndCheckPackets()
177 EXPECT_TRUE(packetizer_->NextPacket(&packet_)); in CreateParseAndCheckPacketsLayers()
219 EXPECT_FALSE(packetizer_->NextPacket(&packet_)); in TEST_F()
337 EXPECT_FALSE(packetizer_->NextPacket(&packet_)); in TEST_F()
492 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST_F()
494 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST_F()
516 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST_F()
518 ASSERT_TRUE(packetizer.NextPacket(&packet)); in TEST_F()
543 ASSERT_TRUE(packetizer0.NextPacket(&packet)); in TEST_F()
545 ASSERT_TRUE(packetizer0.NextPacket(&packet)); in TEST_F()
[all …]
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| D | ulpfec_generator_unittest.cc | 120 packet_generator_.NextPacket(i, 10); in TEST_F()
158 packet_generator_.NextPacket(i * kNumPackets + j, 10); in TEST_F()
186 packet_generator_.NextPacket(i, 10); in TEST_F()
197 packet_generator_.NextPacket(kUlpfecMaxMediaPackets, 10); in TEST_F()
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| D | rtp_format.h | 53 virtual bool NextPacket(RtpPacketToSend* packet) = 0;
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| D | rtp_format_vp9.h | 51 bool NextPacket(RtpPacketToSend* packet) override;
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| D | rtp_format_vp8.h | 58 bool NextPacket(RtpPacketToSend* packet) override;
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| D | rtp_format_video_generic.h | 56 bool NextPacket(RtpPacketToSend* packet) override;
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| D | rtp_packetizer_av1.h | 33 bool NextPacket(RtpPacketToSend* packet) override;
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| D | rtp_format_h264.h | 45 bool NextPacket(RtpPacketToSend* rtp_packet) override;
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| /external/webrtc/test/ |
| D | rtp_file_reader_unittest.cc | 36 while (rtp_packet_source_->NextPacket(&packet)) { in CountRtpPackets()
76 while (rtp_packet_source_->NextPacket(&packet)) { in CountRtpPackets()
86 while (rtp_packet_source_->NextPacket(&packet)) { in CountRtpPacketsPerSsrc()
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| D | rtp_file_reader.h | 45 virtual bool NextPacket(RtpPacket* packet) = 0;
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| /external/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| D | common_header_unittest.cc | 51 EXPECT_EQ(buffer + sizeof(buffer), header.NextPacket()); in TEST()
80 EXPECT_EQ(buffer + sizeof(buffer), header.NextPacket()); in TEST()
88 EXPECT_EQ(buffer + sizeof(buffer), header.NextPacket()); in TEST()
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| D | common_header.h | 39 const uint8_t* NextPacket() const { in NextPacket() function
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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
101 std::unique_ptr<Packet> packet = NextPacket(kTimeoutMs); in CheckNextPacket()
136 return NextPacket(kNoPacketTimeoutMs) == nullptr; in CheckNoPacket()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | rtp_file_source.cc | 54 std::unique_ptr<Packet> RtpFileSource::NextPacket() { in NextPacket() function in webrtc::test::RtpFileSource
57 if (!rtp_reader_->NextPacket(&temp_packet)) { in NextPacket()
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| D | packet_source.h | 31 virtual std::unique_ptr<Packet> NextPacket() = 0;
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| D | constant_pcm_packet_source.h | 34 std::unique_ptr<Packet> NextPacket() override;
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| D | rtpcat.cc | 41 while (input->NextPacket(&packet)) in main()
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| D | rtc_event_log_source.h | 46 std::unique_ptr<Packet> NextPacket() override;
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| D | rtp_file_source.h | 48 std::unique_ptr<Packet> NextPacket() override;
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_receive_test.cc | 95 for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet; in Run()
96 packet = packet_source_->NextPacket()) { in Run()
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| /external/webrtc/modules/video_coding/ |
| D | loss_notification_controller_unittest.cc | 59 Packet NextPacket() { in NextPacket() function in webrtc::__anonfb6c07ec0111::PacketStreamCreator
326 const auto key_frame_packet = packet_stream.NextPacket(); in TEST_P()
334 packet_stream.NextPacket(); in TEST_P()
337 auto repeated_packet = packet_stream.NextPacket(); in TEST_P()
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| /external/webrtc/modules/video_coding/test/ |
| D | stream_generator.h | 47 bool NextPacket(VCMPacket* packet);
|