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Searched refs:file_source (Results 1 – 3 of 3) sorted by relevance

/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
Drtp_analyze.cc66 rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source( in main() local
68 assert(file_source.get()); in main()
73 file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, in main()
79 file_source->RegisterRtpHeaderExtension( in main()
109 packet.reset(file_source->NextPacket()); in main()
Dneteq_rtpplay.cc414 rtc::scoped_ptr<webrtc::test::PacketSource> file_source; in main() local
419 file_source.reset(webrtc::test::RtpFileSource::Create(argv[1])); in main()
422 file_source.reset(event_log_source); in main()
425 assert(file_source.get()); in main()
431 file_source->SelectSsrc(ssrc); in main()
444 rtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket()); in main()
512 next_packet.reset(file_source->NextPacket()); in main()
585 webrtc::test::Packet* temp_packet = file_source->NextPacket(); in main()
/external/webrtc/webrtc/voice_engine/test/cmd_test/
Dvoe_cmd_test.cc653 int file_source = 1; in RunTest() local
658 ASSERT_EQ(1, scanf("%i", &file_source)); in RunTest()
659 if (file_source == 1) { in RunTest()
673 if (file_source == 1) { in RunTest()