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Searched refs:RtpFileSource (Results 1 – 6 of 6) sorted by relevance

/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
Drtp_file_source.cc29 RtpFileSource* RtpFileSource::Create(const std::string& file_name) { in Create()
30 RtpFileSource* source = new RtpFileSource(); in Create()
35 bool RtpFileSource::ValidRtpDump(const std::string& file_name) { in ValidRtpDump()
41 bool RtpFileSource::ValidPcap(const std::string& file_name) { in ValidPcap()
47 RtpFileSource::~RtpFileSource() { in ~RtpFileSource()
50 bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type, in RegisterRtpHeaderExtension()
56 Packet* RtpFileSource::NextPacket() { in NextPacket()
85 RtpFileSource::RtpFileSource() in RtpFileSource() function in webrtc::test::RtpFileSource
89 bool RtpFileSource::OpenFile(const std::string& file_name) { in OpenFile()
Drtp_file_source.h31 class RtpFileSource : public PacketSource {
35 static RtpFileSource* Create(const std::string& file_name);
41 virtual ~RtpFileSource();
55 RtpFileSource();
62 RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
Drtp_analyze.cc66 rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source( in main()
67 webrtc::test::RtpFileSource::Create(argv[1])); in main()
Dneteq_rtpplay.cc416 if (webrtc::test::RtpFileSource::ValidRtpDump(argv[1]) || in main()
417 webrtc::test::RtpFileSource::ValidPcap(argv[1])) { in main()
419 file_source.reset(webrtc::test::RtpFileSource::Create(argv[1])); in main()
/external/webrtc/webrtc/modules/audio_coding/acm2/
Daudio_coding_module_unittest_oldapi.cc903 rtc::scoped_ptr<test::RtpFileSource> packet_source( in Run()
904 test::RtpFileSource::Create(input_file_name)); in Run()
/external/webrtc/webrtc/modules/audio_coding/neteq/
Dneteq_unittest.cc323 rtc::scoped_ptr<test::RtpFileSource> rtp_source_;
406 rtp_source_.reset(test::RtpFileSource::Create(rtp_file)); in OpenInputFile()