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  /external/webrtc/webrtc/modules/audio_coding/neteq/tools/
rtp_analyze.cc 66 rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
68 assert(file_source.get());
73 file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
79 file_source->RegisterRtpHeaderExtension(
109 packet.reset(file_source->NextPacket());
neteq_rtpplay.cc 414 rtc::scoped_ptr<webrtc::test::PacketSource> file_source; local
419 file_source.reset(webrtc::test::RtpFileSource::Create(argv[1]));
422 file_source.reset(event_log_source);
425 assert(file_source.get());
431 file_source->SelectSsrc(ssrc);
444 rtc::scoped_ptr<webrtc::test::Packet> packet(file_source->NextPacket());
512 next_packet.reset(file_source->NextPacket());
585 webrtc::test::Packet* temp_packet = file_source->NextPacket();
  /system/bt/btif/src/
btif_config.cc 52 #define FILE_SOURCE "FileSource"
135 std::string file_source; local
144 file_source = "Backup";
152 file_source = "Legacy";
160 file_source = "Empty";
163 if (!file_source.empty())
164 config_set_string(config, INFO_SECTION, FILE_SOURCE, file_source.c_str());
527 config_get_string(config, INFO_SECTION, FILE_SOURCE, "Original"));
  /external/webrtc/webrtc/voice_engine/test/cmd_test/
voe_cmd_test.cc 653 int file_source = 1; local
658 ASSERT_EQ(1, scanf("%i", &file_source));
659 if (file_source == 1) {
673 if (file_source == 1) {

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