| /external/webrtc/webrtc/test/testsupport/ |
| D | packet_reader_unittest.cc | 44 EXPECT_EQ(-1, reader_->NextPacket(&data_pointer)); in TEST_F()
50 ASSERT_EQ(0, reader_->NextPacket(&packet_data_pointer_)); in TEST_F()
61 int length_to_read = reader_->NextPacket(&data_pointer); in TEST_F()
66 length_to_read = reader_->NextPacket(&data_pointer); in TEST_F()
78 int length_to_read = reader_->NextPacket(&data_pointer); in TEST_F()
83 length_to_read = reader_->NextPacket(&data_pointer); in TEST_F()
93 int length_to_read = reader_->NextPacket(&packet_data_pointer_); in TEST_F()
97 length_to_read = reader_->NextPacket(&packet_data_pointer_); in TEST_F()
101 length_to_read = reader_->NextPacket(&packet_data_pointer_); in TEST_F()
106 length_to_read = reader_->NextPacket(&packet_data_pointer_); in TEST_F()
[all …]
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| D | packet_reader.h | 41 virtual int NextPacket(uint8_t** packet_pointer);
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| 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/ |
| D | packet_buffer_unittest.cc | 31 Packet* NextPacket(int payload_size_bytes);
52 Packet* PacketGenerator::NextPacket(int payload_size_bytes) { in NextPacket() function in webrtc::PacketGenerator
93 Packet* packet = gen.NextPacket(payload_len); in TEST()
116 Packet* packet = gen.NextPacket(payload_len); in TEST()
137 Packet* packet = gen.NextPacket(payload_len); in TEST()
146 Packet* packet = gen.NextPacket(payload_len); in TEST()
166 Packet* packet = gen.NextPacket(payload_len); in TEST()
202 Packet* packet = gen.NextPacket(payload_len); in TEST()
206 Packet* packet = gen.NextPacket(payload_len); in TEST()
270 Packet* packet = gen.NextPacket(kPayloadLength); in TEST()
[all …]
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_format_h264_unittest.cc | 94 ASSERT_TRUE(packetizer->NextPacket(packet.get(), &length, &last)); in TestFua()
100 EXPECT_FALSE(packetizer->NextPacket(packet.get(), &length, &last)); in TestFua()
165 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last)); in TEST()
170 EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last)); in TEST()
195 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last)); in TEST()
199 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last)); in TEST()
204 EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last)); in TEST()
232 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last)); in TEST()
240 EXPECT_FALSE(packetizer->NextPacket(packet, &length, &last)); in TEST()
267 ASSERT_TRUE(packetizer->NextPacket(packet, &length, &last)); in TEST()
[all …]
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| D | producer_fec_unittest.cc | 120 RtpPacket* rtp_packet = generator_->NextPacket(i, 10); 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 | fec_test_helper.cc | 28 RtpPacket* FrameGenerator::NextPacket(int offset, size_t length) { in NextPacket() function in webrtc::FrameGenerator
67 RtpPacket* red_packet = NextPacket(0, packet->length + 1); in BuildFecRedPacket()
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| D | fec_test_helper.h | 37 RtpPacket* NextPacket(int offset, size_t length);
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| D | rtp_format.h | 42 virtual bool NextPacket(uint8_t* buffer,
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| D | rtp_format_video_generic.h | 44 bool NextPacket(uint8_t* buffer,
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| D | rtp_format_vp9.h | 58 bool NextPacket(uint8_t* buffer,
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| /external/webrtc/webrtc/test/ |
| D | rtp_file_reader_unittest.cc | 35 while (rtp_packet_source_->NextPacket(&packet)) { in CountRtpPackets()
75 while (rtp_packet_source_->NextPacket(&packet)) { in CountRtpPackets()
85 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/webrtc/base/ |
| D | testclient.cc | 54 TestClient::Packet* TestClient::NextPacket(int timeout_ms) { in NextPacket() function in rtc::TestClient
91 Packet* packet = NextPacket(kTimeoutMs); in CheckNextPacket()
103 Packet* packet = NextPacket(kNoPacketTimeoutMs); in CheckNoPacket()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | rtp_file_source.cc | 56 Packet* RtpFileSource::NextPacket() { in NextPacket() function in webrtc::test::RtpFileSource
59 if (!rtp_reader_->NextPacket(&temp_packet)) { in NextPacket()
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| D | packet_source.h | 31 virtual Packet* NextPacket() = 0;
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| D | constant_pcm_packet_source.h | 36 Packet* NextPacket() override;
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| D | rtp_file_source.h | 48 Packet* NextPacket() override;
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| D | rtc_event_log_source.h | 46 Packet* NextPacket() override;
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| D | rtpcat.cc | 41 while (input->NextPacket(&packet)) in main()
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| D | constant_pcm_packet_source.cc | 38 Packet* ConstantPcmPacketSource::NextPacket() { in NextPacket() function in webrtc::test::ConstantPcmPacketSource
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
| D | bwe_rtp_play.cc | 62 if (!rtp_reader->NextPacket(&packet)) { in main()
88 if (!rtp_reader->NextPacket(&packet)) { in main()
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| D | acm_receive_test_oldapi.cc | 154 for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet; in Run()
155 packet.reset(packet_source_->NextPacket())) { in Run()
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| D | acm_send_test_oldapi.h | 50 Packet* NextPacket();
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| /external/webrtc/webrtc/modules/video_coding/test/ |
| D | stream_generator.h | 46 bool NextPacket(VCMPacket* packet);
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